Since the core now includes deduplication in the name of CODEC
devices there's no need to add extra for the debugfs directory name.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Since the debugfs directory is current per CODEC we should only init
it when the CODEC is initialised, otherwise we end up with errors
being generated when an attempt is made to add duplicate debugfs
entries.
Since most of this stuff is actually for the card we should refactor
but this can come later.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Gcc complains that ret might be used uninitialized:
sound/usb/format.c: In function ‘snd_usb_parse_audio_format’:
sound/usb/format.c:354: warning: ‘ret’ may be used uninitialized in this function
sound/usb/format.c:354: note: ‘ret’ was declared here
sound/usb/format.c:414: warning: ‘ret’ may be used uninitialized in this function
sound/usb/format.c:414: note: ‘ret’ was declared here
I suppose it could be uninitialized if there is ever a UAC_VERSION_3
released. Anyway this patch is worthwhile if only to silence the gcc
warning.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Any subsequent revisions will have these configuration changes applied
by default.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Change the chip defaults to optimise performance of some of the DSP
functionality.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The commit eb541337b7
ALSA: hda - Make converter setups sticky
changes the semantics of snd_hda_codec_cleanup_stream() not to clean up
the stream at that moment but delay the action. This broke the codes
expecting that the clean-up is done immediately, such as dynamic ADC
changes in some codec drivers.
This patch fixes the issue by introducing a lower helper,
__snd_hda_codec_cleanup_stream(), to allow the immediate clean up.
The original snd_hda_codec_cleanup_stream() is kept as is now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a device is plugged over HDMI, it passes some information in ELD
including the supported PCM parameters like formats, rates, channels.
This patch adds the check to PCM open callback of HDMI streams so that
only valid parameters the device supports are used.
When no device is plugged, the parameters the codec supports are used;
it's mostly all parameters the hardware can work. This is for apps
that are started before device plugging and do probing (e.g. a sound
daemon), so that at least, probing would work even before the device
plugging.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ASoC: add AD1980 obsolete information
ASoC: register cache should be 1 byte aligned for 1 byte long register
ALSA: hda - Adding support for new IDT 92HD87XX codecs
ASoC: Fix inverted mute controls for WM8580
ALSA: HDA: Use model=auto for LG R510
ALSA: hda - Update model entries in HD-Audio-Models.txt
ALSA: hda: document VIA models
ALSA: hda - patch_nvhdmi.c: Add missing codec IDs, unify names
ALSA: hda - add support for Conexant CX20584
ALSA: hda - New snd-hda-intel model/pin config for hp dv7-4000
ALSA: hda - Fix missing stream for second ADC on Realtek ALC260 HDA codec
ALSA: hda - Make converter setups sticky
ALSA: hda - Add support for Acer ZGA ALC271 (1025:047c)
sound/oss: Adjust confusing if indentation
sound: oss: au1550_ac97.c removed duplicated #include
ASoC: Fix for changed Eureka Kconfig symbol names
Nothing should be referencing this any more.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Add back the register restore call, when the codec driver is
removed.
This does not affect normal operation, but it is usefull when
debugging audio through the twl4030 class codecs.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Since the provision of a struct device for the CODEC is now mandatory
we can use container_of() to locate the struct i2c_client and struct
spi_device for relevant devices, removing the need to manually set it
in each driver.
A further patch will automate selection of the control type based on
the bus_type of the struct device, further reducing the amount of
driver code required.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Now soc-cache.c can figure out the I2C and SPI control data from the
device for the CODEC we don't need to manually assign it in drivers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Add support for adding "status = disabled" to an SSI node to incidate that it
is not wired on the board. This replaces the not-so-intuitive previous method
of omitting a codec-handle property.
Signed-off-by: Timur Tabi <timur@freescale.com>
Acked-by: Kumar Gala <galak@kernel.crashing.org>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The error handling code in the OF probe function of the SSI driver is not
freeing all resources correctly.
Since the machine driver no longer calls the DMA driver to provide information
about the SSI, we don't need to keep a list of DMA objects any more. In
addition, the fsl_soc_dma_remove() function is incorrectly removing *all*
DMA objects when it should only remove one.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Update the DMA driver used by the Freescale MPC8610 HPCD audio driver to
support 36-bit physical addresses, for both DMA buffers and the SSI registers.
The DMA driver calls snd_dma_alloc_pages() to allocate the DMA buffers for
playback and capture. This function is just a front-end for
dma_alloc_coherent(). Currently, dma_alloc_coherent() only allocates buffers
in low memory (it ignores GFP_HIGHMEM), so we never actually get a DMA buffer
with a real 36-bit physical address.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The immap_86xx.h header file only defines one data structure: the "global
utilities" register set found on Freescale PowerPC SOCs. Rename this file
to fsl_guts.h to reflect its true purpose, and extend it to cover the "GUTS"
register set on 85xx chips.
Signed-off-by: Timur Tabi <timur@freescale.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Kumar Gala <galak@kernel.crashing.org>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch add sound support for the Goni board based on S5PV210.
The Goni board is based on Samsung SoC(S5PV210) and include
WM8994 codec over I2S to support sound.
The kind of jack is below states :
* SND_JACK_HEADPHONE
* SND_JACK_HEADSET
* SND_JACK_MECHANICAL
: When TV-OUT cable is inserted on Goni board,
the TV-OUT cable isn't connected to television.
* SND_JACK_AVOUT
: When TV-OUT cable is inserted on Goni board,
the TV-OUT cable is connected to television.
Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch add sound support for the Aquila board based on S5PC110.
The Aquila board is based on Samsung SoC(S5PC110) and include
WM8994 codec over I2S to support sound. This uses the I2Sv4 driver
compatible with I2Sv5 to run sound.
The kind of jack is below states :
* SND_JACK_HEADPHONE
* SND_JACK_HEADSET
* SND_JACK_MECHANICAL
: When TV-OUT cable is inserted on Aquila board,
the TV-OUT cable isn't connected to television.
* SND_JACK_AVOUT
: When TV-OUT cable is inserted on Aquila board,
the TV-OUT cable is connected to television.
Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
ASoC: multi-component: SAMSUNG: Fix wrong field name on Aquila board
This patch modify the wrong field name on Aquila board.
Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch extends the ASoC API to allow sound cards to have more than one
CODEC and more than one platform DMA controller. This is achieved by dividing
some current ASoC structures that contain both driver data and device data into
structures that only either contain device data or driver data. i.e.
struct snd_soc_codec ---> struct snd_soc_codec (device data)
+-> struct snd_soc_codec_driver (driver data)
struct snd_soc_platform ---> struct snd_soc_platform (device data)
+-> struct snd_soc_platform_driver (driver data)
struct snd_soc_dai ---> struct snd_soc_dai (device data)
+-> struct snd_soc_dai_driver (driver data)
struct snd_soc_device ---> deleted
This now allows ASoC to be more tightly aligned with the Linux driver model and
also means that every ASoC codec, platform and (platform) DAI is a kernel
device. ASoC component private data is now stored as device private data.
The ASoC sound card struct snd_soc_card has also been updated to store lists
of it's components rather than a pointer to a codec and platform. The PCM
runtime struct soc_pcm_runtime now has pointers to all its components.
This patch adds DAPM support for ASoC multi-component and removes struct
snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec
or runtime PCM level basis rather than using snd_soc_socdev.
Other notable multi-component changes:-
* Stream operations now de-reference less structures.
* close_delayed work() now runs on a DAI basis rather than looping all DAIs
in a card.
* PM suspend()/resume() operations can now handle N CODECs and Platforms
per sound card.
* Added soc_bind_dai_link() to bind the component devices to the sound card.
* Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove
DAI link components.
* sysfs entries can now be registered per component per card.
* snd_soc_new_pcms() functionailty rolled into dai_link_probe().
* snd_soc_register_codec() now does all the codec list and mutex init.
This patch changes the probe() and remove() of the CODEC drivers as follows:-
o Make CODEC driver a platform driver
o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core.
o Removed all static codec pointers (drivers now support > 1 codec dev)
o snd_soc_register_pcms() now done by core.
o snd_soc_register_dai() folded into snd_soc_register_codec().
CS4270 portions:
Acked-by: Timur Tabi <timur@freescale.com>
Some TLV320aic23 and Cirrus platform fixes.
Signed-off-by: Ryan Mallon <ryan@bluewatersys.com>
TI CODEC and OMAP fixes
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Samsung platform and misc fixes :-
Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Reviewed-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
MPC8610 and PPC fixes.
Signed-off-by: Timur Tabi <timur@freescale.com>
i.MX fixes and some core fixes.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
J4740 platform fixes:-
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
CC: Tony Lindgren <tony@atomide.com>
CC: Nicolas Ferre <nicolas.ferre@atmel.com>
CC: Kevin Hilman <khilman@deeprootsystems.com>
CC: Sascha Hauer <s.hauer@pengutronix.de>
CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
CC: Kuninori Morimoto <morimoto.kuninori@renesas.com>
CC: Daniel Gloeckner <dg@emlix.com>
CC: Manuel Lauss <mano@roarinelk.homelinux.net>
CC: Mike Frysinger <vapier.adi@gmail.com>
CC: Arnaud Patard <apatard@mandriva.com>
CC: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This codec has been obsoleted by ADI, so add appropriate warnings to the
source tree to dissuade people from using in new designs based on driver
support.
Signed-off-by: Sonic Zhang <sonic.zhang@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Added the entries for 92HD87B1/3 and 92HD87B2/4 codecs.
These are compatible with existing 83xxx codecs.
Signed-off-by: Charles Chin <Charles.Chin@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Should use Capture rather than ADC so the UI tools can identify their
function more readily.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Sadly these aren't soft controllable and can't be read back either :(
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Otherwise debugfs gets upset when we try to create filenames with /
in them.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Two users report model=auto is needed to make the internal mic work properly.
BugLink: https://bugs.launchpad.net/bugs/495134
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Add missing codec IDs.
* Modify some existing codec names for discrete GPUs to match newly
added IDs. Note: existing names were a mixture of marketing and
engineering GPU names. Equally, there's no reason that codec IDs
have to be specific to a particular GPU or board, so identify
codecs in a less marketing-oriented fashion.
* Reformat codec ID table so it's easier to read, for me at least.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Conexant CX20584 with 141f:5068 seems compatible with other
cxt5066 code. Just add the missing id.
Tested-by: Cristopher Camacho Leandro <ccamacho@linuxmail.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This provides a new model and pin config for the snd-hda-intel
92HD83XXX codec for hp laptop model dv7-4000, enabling the subwoofer.
Signed-off-by: Steven Eastland <seastland at gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I discovered tonight that ALSA no longer sets up a stream for the second ADC
provided by the Realtek ALC260 HDA codec. At some point alc_build_pcms()
started using stream_analog_alt_capture when constructing the second ADC
stream, but patch_alc260() was never updated accordingly. I have no idea
when this regression occurred. The trivial patch to patch_alc260() given
below fixes the problem as far as I can tell. The patch is against 2.6.35.
Signed-off-by: Jonathan Woithe <jwoithe@physics.adelaide.edu.au>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (214 commits)
ALSA: hda - Add pin-fix for HP dc5750
ALSA: als4000: Fix potentially invalid DMA mode setup
ALSA: als4000: enable burst mode
ALSA: hda - Fix initial capsrc selection in patch_alc269()
ASoC: TWL4030: Capture route runtime DAPM ordering fix
ALSA: hda - Add PC-beep whitelist for an Intel board
ALSA: hda - More relax for pending period handling
ALSA: hda - Define AC_FMT_* constants
ALSA: hda - Fix beep frequency on IDT 92HD73xx and 92HD71Bxx codecs
ALSA: hda - Add support for HDMI HBR passthrough
ALSA: hda - Set Stream Type in Stream Format according to AES0
ALSA: hda - Fix Thinkpad X300 so SPDIF is not exposed
ALSA: hda - FIX to not expose SPDIF on Thinkpad X301, since it does not have the ability to use SPDIF
ASoC: wm9081: fix resource reclaim in wm9081_register error path
ASoC: wm8978: fix a memory leak if a wm8978_register fail
ASoC: wm8974: fix a memory leak if another WM8974 is registered
ASoC: wm8961: fix resource reclaim in wm8961_register error path
ASoC: wm8955: fix resource reclaim in wm8955_register error path
ASoC: wm8940: fix a memory leak if wm8940_register return error
ASoC: wm8904: fix resource reclaim in wm8904_register error path
...
Call the gpio reset platform function instead of using the flawed
ac97 functionality of the MPC5200(b)
From MPC5200B User's Manual:
"Some AC97 devices goes to a test mode, if the Sync line is high
during the Res line is low (reset phase). To avoid this behavior the
Sync line must be also forced to zero during the reset phase. To do
that, the pin muxing should switch to GPIO mode and the GPIO control
register should be used to control the output lines."
Signed-off-by: Eric Millbrandt <emillbrandt@dekaresearch.com>
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
* git://git.kernel.org/pub/scm/linux/kernel/git/brodo/pcmcia-2.6:
pcmcia: avoid buffer overflow in pcmcia_setup_isa_irq
pcmcia: do not request windows if you don't need to
pcmcia: insert PCMCIA device resources into resource tree
pcmcia: export resource information to sysfs
pcmcia: use struct resource for PCMCIA devices, part 2
pcmcia: remove memreq_t
pcmcia: move local definitions out of include/pcmcia/cs.h
pcmcia: do not use io_req_t when calling pcmcia_request_io()
pcmcia: do not use io_req_t after call to pcmcia_request_io()
pcmcia: use struct resource for PCMCIA devices
pcmcia: clean up cs.h
pcmcia: use pcmica_{read,write}_config_byte
pcmcia: remove cs_types.h
pcmcia: remove unused flag, simplify headers
pcmcia: remove obsolete CS_EVENT_ definitions
pcmcia: split up central event handler
pcmcia: simplify event callback
pcmcia: remove obsolete ioctl
Conflicts in:
- drivers/staging/comedi/drivers/*
- drivers/staging/wlags49_h2/wl_cs.c
due to dev_info_t and whitespace changes
of_device is just an alias for platform_device, so remove it entirely. Also
replace to_of_device() with to_platform_device() and update comment blocks.
This patch was initially generated from the following semantic patch, and then
edited by hand to pick up the bits that coccinelle didn't catch.
@@
@@
-struct of_device
+struct platform_device
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Reviewed-by: David S. Miller <davem@davemloft.net>
So far, we reset the converter setups like the stream-tag, the
channel-id and format-id in prepare callbacks, and clear them in
cleanup callbacks. This often causes a silence of the digital
receiver for a couple of seconds.
This patch tries to delay the converter setup changes as much as
possible. The converter setups are cached and aren't reset as long
as the same values are used. At suspend/resume, they are cleared
to be recovered properly, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Indent the branch of an if.
The semantic match that finds this problem is as follows:
(http://coccinelle.lip6.fr/)
// <smpl>
@r disable braces4@
position p1,p2;
statement S1,S2;
@@
(
if (...) { ... }
|
if (...) S1@p1 S2@p2
)
@script:python@
p1 << r.p1;
p2 << r.p2;
@@
if (p1[0].column == p2[0].column):
cocci.print_main("branch",p1)
cocci.print_secs("after",p2)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'next-devicetree' of git://git.secretlab.ca/git/linux-2.6: (63 commits)
of/platform: Register of_platform_drivers with an "of:" prefix
of/address: Clean up function declarations
of/spi: call of_register_spi_devices() from spi core code
of: Provide default of_node_to_nid() implementation.
of/device: Make of_device_make_bus_id() usable by other code.
of/irq: Fix endian issues in parsing interrupt specifiers
of: Fix phandle endian issues
of/flattree: fix of_flat_dt_is_compatible() to match the full compatible string
of: remove of_default_bus_ids
of: make of_find_device_by_node generic
microblaze: remove references to of_device and to_of_device
sparc: remove references to of_device and to_of_device
powerpc: remove references to of_device and to_of_device
of/device: Replace of_device with platform_device in includes and core code
of/device: Protect against binding of_platform_drivers to non-OF devices
of: remove asm/of_device.h
of: remove asm/of_platform.h
of/platform: remove all of_bus_type and of_platform_bus_type references
of: Merge of_platform_bus_type with platform_bus_type
drivercore/of: Add OF style matching to platform bus
...
Fix up trivial conflicts in arch/microblaze/kernel/Makefile due to just
some obj-y removals by the devicetree branch, while the microblaze
updates added a new file.
The WM8962 is a low power, high performance stereo CODEC designed for
portable digital audio applications.
This initial driver release supports the key audio paths of the WM8962.
Extended functionality, such as microphone detection, digital microphones
and the advanced DSP signal enhancements provided by the device are not
yet supported.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The NID 0x11 on HP dc5750 with ALC260 should be a speaker although BIOS
gives it as a line-out. This patch adds a quirk to fix the pin config
so that the real line-out is used properly.
Reference: bnc#624118
https://bugzilla.novell.com/show_bug.cgi?id=624118
Signed-off-by: Takashi Iwai <tiwai@suse.de>
My previous patch assumed that the DMA mode (represented by 3 lowest bits of
ALS4K_GCR99_DMA_EMULATION_CTRL register) is set to the default value 0. If
that's not the case, it might result in invalid mode to be set.
This patch fixes this potential problem.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/rafael/suspend-2.6:
PM / Runtime: Add runtime PM statistics (v3)
PM / Runtime: Make runtime_status attribute not debug-only (v. 2)
PM: Do not use dynamically allocated objects in pm_wakeup_event()
PM / Suspend: Fix ordering of calls in suspend error paths
PM / Hibernate: Fix snapshot error code path
PM / Hibernate: Fix hibernation_platform_enter()
pm_qos: Get rid of the allocation in pm_qos_add_request()
pm_qos: Reimplement using plists
plist: Add plist_last
PM: Make it possible to avoid races between wakeup and system sleep
PNPACPI: Add support for remote wakeup
PM: describe kernel policy regarding wakeup defaults (v. 2)
PM / Hibernate: Fix typos in comments in kernel/power/swap.c
Enable burst mode to prevent dropouts during high PCI bus usage.
The card is useless in X without this because of dropouts when anything moves
on the screen (at least with PCI VGA card). Enabling this is also recommended
by the datasheet (page 48).
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In patch_alc269(), we initialize the primary capsrc so that the device
works from the beginning. It issues CONNECT_SEL verb no matter which
widget is although some widget (e.g. 0x23) has no connection selection
but a mixer, which requires unmuting instead.
This patch fixes the initialization of capsrc by re-using the code as
a helper function.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the ordering problem in DAPM domain, when the user
changes between digital and analog sources during active
capture (or loopback) scenario.
Before this patch, when the user changed from analog source
to digital there were a short time, when the codec enabled
analog mic bias (2.2 volts) instead of the correct digital
mic bias (1.8 volts) to the digital microphones.
This behaviour caused by the former implementation of
selecting the correct type of bias. This was done at the
POST_REG event of the DAPM_MUX_E("TXx Capture Route")
widget.
By moving the bias type selection as DAPM_SUPPLY and
connecting it to the corresponding digimic widget the
problematic situation can be avoided.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
An Intel board needs a white-list entry to enable PC-beep.
Otherwise the driver misdetects (due to bogus BIOS info) and ignores
the PC-beep on 2.6.35.
Reported-and-tested-by: Leandro Lucarella <luca@llucax.com.ar>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the pending periods are often bogus and take long time until
actually processed, it often results in a high CPU usage of the hd-audio
workq. Overall it's better to have low CPU consumption by avoiding a
too tight loop rather than the wake-up timing accuracy.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix HDA beep frequency on IDT 92HD73xx and 92HD71Bxx codecs.
These codecs use the standard beep frequency calculation although the
datasheet says it's linear frequency.
Other IDT/STAC codecs might have the same problem. They should be
fixed individually later.
Signed-off-by: Daniel J Blueman <daniel.blueman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Passing IEC 61937 encapsulated compressed audio at bitrates over 6.144
Mbps (i.e. more than a single 2-channel 16-bit 192kHz IEC 60958 link)
over HDMI requires the use of HBR Audio Stream Packets instead of Audio
Sample Packets.
Enable HBR mode when the stream has 8 channels and the Non-PCM bit is
set.
If the audio converter is not connected to any HBR-capable pins, return
-EINVAL in prepare().
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set bit 15 (Stream Type) of HDA Stream Format to 1 (Non-PCM) when IEC958
channel status bit 1 (AES0 & 0x02) is set to 1 (non-audio).
This is a prequisite for HDMI HBR passthrough.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just as with the X301. The X300 does not have a way to do SPDIF either.
It does not have a dock connector, nor does it have the SPDIF through
the headphone jack.
This patch fixes it so X300 does not show SPDIF, since it cannot do it.
To add all Lenovo Thinkpads had different codec subsytem IDs:
X300:
http://launchpadlibrarian.net/34862838/Card0.Codecs.codec.0.txt
Signed-off-by: Jerone Young <jerone.young@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Lenovo X301 does not have the ability to connect to a docking
station to use the SPDIF port. It also does not have the ability to do
SPDIF though the headphone jack or Display Port jacks.
This patch fixes it so this is not exposed for the X301 and users do
think it has the ability to do SPDIF.
I tested both headphone & display port jacks and it is not there. I have
tested this patch and it works great.
Also to add the other Thinkpads have different subsystem codec IDs.
Here are examples:
X301:
http://launchpadlibrarian.net/31561902/Card0.Codecs.codec.0.txt
X200:
http://launchpadlibrarian.net/49055036/Card0.Codecs.codec.0.txt
W500:
http://launchpadlibrarian.net/36276057/Card0.Codecs.codec.0.txt
Signed-off-by: Jerone Young <jerone.young@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes the error path in wm9081_register to properly free resources.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is a memory leak found if wm8978_register() fail.
This patch moves the buffer allocate and release
at the same level to prevent the memory leak.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Reviewed-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
wm8974 is allocated in wm8974_i2c_probe() but is not freed if wm8974_register()
return -EINVAL (if another WM8974 is registered).
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes the error path in wm8961_register to properly free resources.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes the error path in wm8955_register to properly free resources.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds checking for wm8940_register return value,
and does kfree(wm8940) if wm8940_register() fail.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch includes below fixes:
1. wm8904 need to be kfreed in wm8904_register() error path before return.
2. fix the error path for snd_soc_register_codec() fail and
snd_soc_register_dai() fail to properly free resources.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
wm8711 is allocated in either wm8711_spi_probe() or wm8711_i2c_probe() but is
not freed if wm8711_register() return -EINVAL(if another ad1836 is registered).
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch includes below fixes:
1. If another WM8523 is registered, need to kfree wm8523 before return -EINVAL.
2. If snd_soc_register_codec failed, goto error path to properly free resources.
3. Instead of using mixed in-line and goto style cleanup, use goto style error
handling if snd_soc_register_dai failed.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
da7210 should be kfreed if da7210_init() return error.
This patch also fixes the error handing in the case of snd_soc_register_dai()
fail by adding snd_soc_unregister_codec() in error path.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ak4642 should be kfreed if ak4642_init() return error.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Reviewed-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ad1836 is allocated in ad1836_spi_probe() but is not freed if ad1836_register()
return -EINVAL (if another ad1836 is registered).
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Barry Song <21cnbao@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8741 is a very high performance stereo DAC designed for audio
applications such as professional recording systems, A/V receivers and
high specification CD, DVD and home theatre systems. The device supports
PCM data input word lengths from 16 to 32-bits and sampling rates up to
192kHz. The WM8741 also supports DSD bit-stream data format, in both
direct DSD and PCM-converted DSD modes.
TODO: Expand wm8741_set_dai_sysclk and rate_constraint members to
allow for all supported sample rate / Master Clock frequency combinations.
Fully enable control of supplies.
Signed-off-by: Ian Lartey <ian@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The use of sDMA packet mode in THRESHOLD mode removes the restriction on the
period size.
With the extended THRESHOLD mode user space can ask for any
period size it wishes, and the driver will configure the
sDMA and McBSP FIFO accordingly.
Replace the hw_rule for the period size with static constraint,
which will make sure that the period size will be always
even (to avoid prime period size, which could be possible in
mono stream)
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Utilize the sDMA controller's packet syncronization mode, when
the McBSP FIFO is in use (by extending the THRESHOLD mode).
When the sDMA is configured for packet mode, the sDMA frame size
does not need to match with the McBSP threshold configuration.
Uppon DMA request the sDMA will transfer packet size number of
words, and still trigger interrupt on frame boundary.
The patch extends the original THRESHOLD mode by doing the
following:
if (period_words <= max_threshold)
Current THRESHOLD mode configuration
Otherwise (period_words > max_threshold)
McBSP threshold = sDMA packet size
sDMA frame size = period size
With the extended THRESHOLD mode we can remove the constraint
for the maximum period size, since if the period size is
bigger than the maximum allowed threshold, than the driver
will switch to packet mode, and picks the best (biggest)
threshold value, which can divide evenly the period size.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
To make the code a bit more readable, change the indexed
references to the omap_mcbsp_dai_dma_params elements with
pointer.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
In preparation for the extended threshold mode (sDMA packet mode
support), the code need to be restructured.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Match usb ids in usb/quirks-table.h for some Hauppage HVR-950Q models
and for the HVR850 model to those ids at the end of au0828-cards.c
Thanks to nhJm449 for pointing out the problem.
Signed-off-by: John S Gruber <JohnSGruber@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current ALC259/268/269 parser ignores some pins as unhandled,
but user won't notice what goes wrong. So, added a warning message
for the ignored pins as a hint.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Call snd_hda_shutup_pins() for power-saving and reboot-notifier in
patch_conexant.c as well as other codecs. This will reduce the pop
noise in power-save mode.
Reference: bnc#624896
https://bugzilla.novell.com/show_bug.cgi?id=624896
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If BIOS sets up the input pin as VREF 50, use the value as is instead of
overriding forcibly to VREF 80. This fixes the quality of inputs on
some devices like Packard-Bell M5210.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since ALC259/269 use the same parser of ALC268, the pin 0x1b was ignored
as an invalid widget. Just add this NID to handle properly.
This will add the missing mixer controls for some devices.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make a helper function to parse the digital I/Os of all Realtek codecs
to simplify the code and to ensure the setups.
Also, initialize digital I/O pins properly in init callbacks. Some BIOS
seem to leave pins uninitialized.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some ALC662-compatible codecs like ALC892 may have more than 4
connections for the input source. Use HDA_MAX_CONNECTIONS instead of
the fixed magic number 4.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Add a PC-beep workaround for ASUS P5-V
ALSA: hda - Assume PC-beep as default for Realtek
ALSA: hda - Don't register beep input device when no beep is available
ALSA: hda - Fix pin-detection of Nvidia HDMI
Current FSI driver id is not only 0
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The non-standard name "iMic" makes PulseAudio ignore the microphone.
BugLink: https://launchpad.net/bugs/605101
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASUS P5-V provides a SSID that unexpectedly matches with the value
compilant with Realtek's specification. Thus the driver interprets
it badly, resulting in non-working PC beep.
This patch adds a white-list for such a case; a white-list of known
devices with working PC beep.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
get_user() may fail, if so return -EFAULT.
[Fixed one missing place by tiwai]
Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Platform parameter to enable automatic FIFO configuration when
the codec is in Mode1 or Mode7 FIFO mode.
When this mode is selected, the controls for changing
nSample (in Mode1), and UTHR (in Mode7) are not added.
The driver configures the FIFO configuration based on
the stream's period size in a way, that every burst will
read period size of data from the host.
In Mode7 we need to use a formula, which gives close enough
aproximation for the burst length from the host point
of view.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Replace the hardwired latency definition with platform data
parameter, and simplify the nSample parameter calculation.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Enable PC-beep as default for hardwares that aren't compliant with the
SSID value Realtek requires. In such a case, better to enable the beep
to avoid a regression.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We check now the availability of PC beep and skip the build of beep
mixers, but the driver still registers the input device. This should
be checked as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The behavior of Nvidia HDMI codec regarding the pin-detection unsol events
is based on the old HD-audio spec, i.e. PD bit indicates only the update
and doesn't show the current state. Since the current code assumes the
new behavior, the pin-detection doesn't work relialby with these h/w.
This patch adds a flag for indicating the old spec, and fixes the issue
by checking the pin-detection explicitly for such hardware.
Tested-by: Wei Ni <wni@nvidia.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The correct size should be sizeof(gRESP_HPI_SUBSYS_FIND_ADAPTERS),
sizeof(&gRESP_HPI_SUBSYS_FIND_ADAPTERS) is incorrect.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is no need to handle POST_PMU, POST_PMD event with
the Capture Route widget.
It is enough to handle POST_REG event, since that will come
when the user changes the routing, and we will switch the needed
bits in the registers.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
WARNING: sound/soc/au1x/snd-soc-au1xpsc-i2s.o(.data+0xa8): Section mismatch in reference from the variable au1xpsc_i2s_driver to the function .init.text:au1xpsc_i2s_drvprobe()
The variable au1xpsc_i2s_driver references
the function __init au1xpsc_i2s_drvprobe()
If the reference is valid then annotate the
variable with __init* or __refdata (see linux/init.h) or name the variable:
*_template, *_timer, *_sht, *_ops, *_probe, *_probe_one, *_console,
Signed-off-by: Ralf Baechle <ralf@linux-mips.org>
The commit afbd9b8448
ALSA: hda - Limit the amp value to write
introduced a regression for codec setups with amp offsets like IDT/STAC
codecs. The limit value should be a raw value without offset calculation.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Both of_bus_type and of_platform_bus_type are just #define aliases
for the platform bus. This patch removes all references to them and
switches to the of_register_platform_driver()/of_unregister_platform_driver()
API for registering.
Subsequent patches will convert each user of of_register_platform_driver()
into plain platform_drivers without the of_platform_driver shim. At which
point the of_register_platform_driver()/of_unregister_platform_driver()
functions can be removed.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: David S. Miller <davem@davemloft.net>
HeadPhone Playback Volume control register of DA7210 has
reserved area. This patch considered it as mute.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'fix/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ASoC: Select wm_hubs automatically for WM8994
ASoC: Remove duplicate AUX definition from WM8776
ASoC:: remove a redundant snd_soc_unregister_codec call in wm8988_register
ASoC: wm8727: add a missing return in wm8727_platform_probe
ASoC: fsi: fixup wrong value setting order of TDM
ASoC: fsi: fixup clock inversion operation
CONFIG_SOFTOSS* doesn't exist in Kconfig or somewhere
else, therefore removing all references for it from the source code.
Signed-off-by: Christian Dietrich <qy03fugy@stud.informatik.uni-erlangen.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When digital microphones are connected to twl, delay is
needed after enabling the digimic interface of the codec.
Add new parameter for the setup data, which can be used
to pass the apropriate delay in ms after the digimic
interface has been enabled.
Without certain delay (in certain HW configuration) the
beggining of the recorded sample contains a glitch, which
is generated by the digital microphones.
Delaying the micbias1, 2 (which is the bias for the digimic0
or 1) does not help, since the glitch is coming after
switching the digimic interface.
Reversing the micbias and digimic enable order does not
work either (in that case the wait need to be added after
the micbias enabled).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
AIF1ADC TDM mode has no effect other than causing the ADCDAT line to
be tristated rather than driven low on clock cycles where there is no
data to be transmitted. If the clock cycle is idle then there should
be no devices using the data so tristating should have no adverse
effects.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Currently the EDMA queue to be used by for servicing ASP through
internal RAM is fixed to EDMAQ_0 and that to service internal RAM
from external RAM is fixed to EDMAQ_1.
This may not be the desirable configuration on all platforms. For
example, on DM365, queue 0 has large fifo size and is more suitable
for video transfers. Having audio and video transfers on the same
queue may lead to starvation on audio side.
platform data as defined currently passes a queue number to the driver
but that remains unused inside the driver.
Fix this by defining one queue each for ASP and RAM transfers in the
platform data and using it inside the driver.
Since EDMAQ_0 maps to 0, thats the queue that will be used if
the asp queue number is not initialized. None of the platforms
currently utilize ping-pong transfers through internal RAM so that
functionality remains unchanged too.
This patch has been tested on DM644x and OMAP-L138 EVMs.
Signed-off-by: Sekhar Nori <nsekhar@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch modify I2Sv2 driver to support Samsung SoC(S5PV210).
Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Otherwise all machine drivers need to do so.
Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Eric Bénard <eric@eukrea.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is a follow on patch adds support for AMD based Lenovo G series
machines, such as the Lenovo G555.
Signed-off-by: Jerone Young <jerone.young@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In situation when appl_ptr is far greater then hw_ptr, the hw_avail value
can be greater than buffer_size. Check for this.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The function IDs are different for audio and modem. Do not mix them.
Also, show the unsolicited bit in the function_id register.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
All current users of pm_qos_add_request() have the ability to supply
the memory required by the pm_qos routines, so make them do this and
eliminate the kmalloc() with pm_qos_add_request(). This has the
double benefit of making the call never fail and allowing it to be
called from atomic context.
Signed-off-by: James Bottomley <James.Bottomley@suse.de>
Signed-off-by: mark gross <markgross@thegnar.org>
Signed-off-by: Rafael J. Wysocki <rjw@sisk.pl>
If kzalloc() fails we must exit with -ENOMEM. Also we must free
allocated runtime->private_data on error as it would be lost on next
call to snd_imx_open().
Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If kzalloc() fails we must exit with -ENOMEM. Also we must free
allocated runtime->private_data on error as it would be lost on next
call to snd_imx_open().
Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The code checks 'davinci_vc' after kzalloc() and do not checks
'davinci_vcif_dev' that kzalloc() result is assigned to. It seems that
it is a typo (autocompletion?).
Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Specified ID is necessary, when some codecs are used with FSI.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In order to reduce pop-noise at powering up/down of the DACs and Drivers,
these components have to be handled in a specific sequence. Headset,
Handsfree, and Earphone drivers are now registered as PGA components to
ensure DACs are enabled first.
Also, add a delay to leave time for DACs to settle before
continuing power up/down sequence.
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
bytes_per_sec is unsigned, so if snd_pcm_format_width() return error we
would not see it.
Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The description has been expanded to explain the time-out
value provided by the power_save module parameter.
Signed-off-by: Michael Witten <mfwitten@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a device is powered down volatile registers can't be read so
attempts to display codec_reg will show error values, and obviously
it is also possible for there to be hardware errors too. Check for
errors from reads and display them more clearly when formatting
codec_reg.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
snd_soc_unregister_codec is called twice if snd_soc_register_dai fail.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
otherwise the error path will always be executed.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This broke in sound/oss: convert to unlocked_ioctl, when I missed one
of the ioctl functions still using the inode pointer.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some VAIO models with ALC275 have dual ADCs for both internal and external
mics, and the driver needs to switch one of them appropriately.
This patch adds a basic support for this functionality, dynamic switching
between two ADCs per jack plug state.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Annotate platform probe callback with __devinit instead of plain __init.
Signed-off-by: Manuel Lauss <manuel.lauss@googlemail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The controller has mute/unmute capability and some bootloader may mute
them at boot. If it's not handled, all things will seem to be working
but no sound will come out of the speaker/headphone.
Signed-off-by: Arnaud Patard <arnaud.patard@rtp-net.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Kirkwood controller needs to be informed if the audio stream is mono
or not. Failing to do so will result in playing at the wrong speed.
Signed-off-by: Arnaud Patard <arnaud.patard@rtp-net.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch add hw_params to snd_soc_dai_ops,
because board specific set_rate is needed
when FSI was used as master mode.
This patch remove fsi_clk_ctrl from fsi_dai_startup,
because clock should be disabled before set_rate.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch didn't use dev_err,
because it is difficult to get struct device here.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Many registers which were grouped by category were added in FSI2.
To make easy to switch FSI/FSI2, fsi_core was added instead of fsi_regs.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Joe Perches <joe@perches.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Joe Perches <joe@perches.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
channel size should be set before setting register value
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Clock inversion should be specified by each flags bit.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Restructure the DAPM connections in order to enable
only the needed DAC (out of four in twl4030 series).
I need to keep the 'AIF Enable' supply connected to
the L2/R2 digital path, since the digital loopback
needs AIF and APLL running.
If no valid route available, than none of the DAC will
be powered, but the AIF and APLL is going to be enabled.
Furthermore, if only one audio path have valid route,
than only the corresponding DAC will be powered.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsomicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
When the gain is configured using dB value it was
not possible to use -24dB since the loopback got
muted instead of -24dB.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsomicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
These are the final conversions for the ioctl file operation so we can remove
it in the next merge window.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This moves the lock_kernel() call from soundcore_open
to the individual OSS device drivers, where we can deal
with it one driver at a time if needed, or just kill
off the drivers.
All core components in ALSA already provide
adequate locking in their open()-functions
and do not require the big kernel lock, so
there is no need to add the BKL there.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Correctly configure bidirectional pins when resuming; do not power down
widgets when they are needed for Smart5.1 output; and on 3-jack boards,
create the streams and controls needed for six channels.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-and-tested-by: Viliam Kubis <viliam.kubis@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As per-stream volume controls, the DXS controls are not intended to
adjust the overall sound level and so are initialized every time
a stream is opened. However, there are special situations where one
wants to reduce the overall volume in the digital domain, i.e., before
the AC'97 codec's PCM volume control. To allow this, add a module
parameter that sets the initial DXS volume.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Tested-by: Soeren D. Schulze <soeren.d.schulze@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It is not advisable to print a warning when a device does not support
setting the sample rate because this is perfectly valid for devices with
a single rate or where rates are implicitly changed by selecting another
alternate setting.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check the amp max value at put callbacks and set the upper limit
so that the driver won't write any invalid value over the defined
range.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the beep mixer controls to Conexant codecs.
They simply control the digital beep generator widget.
For cx5047, I couldn't find any beep generator, so it's not implemented
there.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Many codecs now clear the pin controls at suspend via snd_hda_shutup_pins()
for reducing the click noise at power-off. But this leaves some pins
uninitialized, and they'll be never recovered after resume.
This patch adds the proper recovery of cleared pin controls on resume.
Also it adds a check of bus->shutdown so that pins won't be cleared at
module unloading.
Reference: Kernel bug 16339
http://bugzilla.kernel.org/show_bug.cgi?id=16339
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
