The beep_mode=2 option was introduced to make the beep mixer
controlling the beep input allocation/deallocation dynamically, so
that a user can switch between HD-audio codec digital beep and the
system beep only via mixer API. This was necessary because the
keyboard driver took only the first input beep instance at that time.
However, the recent keyboard driver already processes the multiple
input instances, thus there is no point to keep this mode.
Let's remove it.
Acked-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is the first big chunk for 3.5 merges of sound stuff.
There are a few big changes in different areas. First off, the
streaming logic of USB-audio endpoints has been largely rewritten
for the better support of "implicit feedback". If anything about USB
got broken, this change has to be checked.
For HD-audio, the resume procedure was changed; instead of delaying
the resume of the hardware until the first use, now waking up immediately
at resume. This is for buggy BIOS.
For ASoC, dynamic PCM support and the improved support for digital links
between off-SoC devices are major framework changes.
Some highlights are below:
* HD-audio
- Avoid the accesses of invalid pin-control bits that may stall the codec
- V-ref setup cleanups
- Fix the races in power-saving code
- Fix the races in codec cache hashes and connection lists
- Split some common codes for BIOS auto-parser to hda_auto_parser.c
- Changed the PM resume code to wake up immediately for buggy BIOS
- Creative SoundCore3D support
- Add Conexant CX20751/2/3/4 codec support
* ASoC
- Dynamic PCM support, allowing support for SoCs with internal routing
through components with tight sequencing and formatting constraints
within their internal paths or where there are multiple components
connected with CPU managed DMA controllers inside the SoC.
- Greatly improved support for direct digital links between off-SoC
devices, providing a much simpler way of connecting things like digital
basebands to CODECs.
- Much more fine grained and robust locking, cleaning up some of the
confusion that crept in with multi-component.
- CPU support for nVidia Tegra 30 I2S and audio hub controllers and
ST-Ericsson MSP I2S controolers
- New CODEC drivers for Cirrus CS42L52, LAPIS Semiconductor ML26124, Texas
Instruments LM49453.
- Some regmap changes needed by the Tegra I2S driver.
- mc13783 audio support.
* Misc
- Rewrite with module_pci_driver()
- Xonar DGX support for snd-oxygen
- Improvement of packet handling in snd-firewire driver
- New USB-endpoint streaming logic
- Enhanced M-audio FTU quirks and relevant cleanups
- Increment the support of OSS devices to 256
- snd-aloop accuracy improvement
There are a few more pending changes for 3.5, but they will be
sent slightly later as partly depending on the changes of DRM.
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Merge tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This is the first big chunk for 3.5 merges of sound stuff.
There are a few big changes in different areas. First off, the
streaming logic of USB-audio endpoints has been largely rewritten for
the better support of "implicit feedback". If anything about USB got
broken, this change has to be checked.
For HD-audio, the resume procedure was changed; instead of delaying
the resume of the hardware until the first use, now waking up
immediately at resume. This is for buggy BIOS.
For ASoC, dynamic PCM support and the improved support for digital
links between off-SoC devices are major framework changes.
Some highlights are below:
* HD-audio
- Avoid accesses of invalid pin-control bits that may stall the codec
- V-ref setup cleanups
- Fix the races in power-saving code
- Fix the races in codec cache hashes and connection lists
- Split some common codes for BIOS auto-parser to hda_auto_parser.c
- Changed the PM resume code to wake up immediately for buggy BIOS
- Creative SoundCore3D support
- Add Conexant CX20751/2/3/4 codec support
* ASoC
- Dynamic PCM support, allowing support for SoCs with internal
routing through components with tight sequencing and formatting
constraints within their internal paths or where there are multiple
components connected with CPU managed DMA controllers inside the
SoC.
- Greatly improved support for direct digital links between off-SoC
devices, providing a much simpler way of connecting things like
digital basebands to CODECs.
- Much more fine grained and robust locking, cleaning up some of the
confusion that crept in with multi-component.
- CPU support for nVidia Tegra 30 I2S and audio hub controllers and
ST-Ericsson MSP I2S controolers
- New CODEC drivers for Cirrus CS42L52, LAPIS Semiconductor ML26124,
Texas Instruments LM49453.
- Some regmap changes needed by the Tegra I2S driver.
- mc13783 audio support.
* Misc
- Rewrite with module_pci_driver()
- Xonar DGX support for snd-oxygen
- Improvement of packet handling in snd-firewire driver
- New USB-endpoint streaming logic
- Enhanced M-audio FTU quirks and relevant cleanups
- Increment the support of OSS devices to 256
- snd-aloop accuracy improvement
There are a few more pending changes for 3.5, but they will be sent
slightly later as partly depending on the changes of DRM."
Fix up conflicts in regmap (due to duplicate patches, with some further
updates then having already come in from the regmap tree). Also some
fairly trivial context conflicts in the imx and mcx soc drivers.
* tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (280 commits)
ALSA: snd-usb: fix stream info output in /proc
ALSA: pcm - Add proper state checks to snd_pcm_drain()
ALSA: sh: Fix up namespace collision in sh_dac_audio.
ALSA: hda/realtek - Fix unused variable compile warning
ASoC: sh: fsi: enable chip specific data transfer mode
ASoC: sh: fsi: call fsi_hw_startup/shutdown from fsi_dai_trigger()
ASoC: sh: fsi: use same format for IN/OUT
ASoC: sh: fsi: add fsi_version() and removed meaningless version check
ASoC: sh: fsi: use register field macro name on IN/OUT_DMAC
ASoC: tegra: Add machine driver for WM8753 codec
ALSA: hda - Fix possible races of accesses to connection list array
ASoC: OMAP: HDMI: Introduce codec
ARM: mx31_3ds: Add sound support
ASoC: imx-mc13783 cleanup
mx31moboard: Add sound support
ASoC: mc13783 codec cleanups
ASoC: add imx-mc13783 sound support
ASoC: Add mc13783 codec
mfd: mc13xxx: add codec platform data
ASoC: don't flip master of DT-instantiated DAI links
...
Pull trivial updates from Jiri Kosina:
"As usual, it's mostly typo fixes, redundant code elimination and some
documentation updates."
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial: (57 commits)
edac, mips: don't change code that has been removed in edac/mips tree
xtensa: Change mail addresses of Hannes Weiner and Oskar Schirmer
lib: Change mail address of Oskar Schirmer
net: Change mail address of Oskar Schirmer
arm/m68k: Change mail address of Sebastian Hess
i2c: Change mail address of Oskar Schirmer
net: Fix tcp_build_and_update_options comment in struct tcp_sock
atomic64_32.h: fix parameter naming mismatch
Kconfig: replace "--- help ---" with "---help---"
c2port: fix bogus Kconfig "default no"
edac: Fix spelling errors.
qla1280: Remove redundant NULL check before release_firmware() call
remoteproc: remove redundant NULL check before release_firmware()
qla2xxx: Remove redundant NULL check before release_firmware() call.
aic94xx: Get rid of redundant NULL check before release_firmware() call
tehuti: delete redundant NULL check before release_firmware()
qlogic: get rid of a redundant test for NULL before call to release_firmware()
bna: remove redundant NULL test before release_firmware()
tg3: remove redundant NULL test before release_firmware() call
typhoon: get rid of redundant conditional before all to release_firmware()
...
Add the PCI ID of the Asus Xonar DGX card; it's otherwise
identical with the DG.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since there are still many Acer models that might not be covered by
the current fixup table, let's add back a few typical model names so
that user can test the fixup without recompiling.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Usage of /etc/modprobe.conf file was deprecated by module-init-tools and
is no longer parsed by new kmod tool. References to this file are
replaced in Documentation, comments and Kconfig according to the
context.
There are also some references to the old /etc/modules.conf from 2.4
kernels that are being removed.
Signed-off-by: Lucas De Marchi <lucas.demarchi@profusion.mobi>
Acked-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Mauro Carvalho Chehab <mchehab@redhat.com>
Signed-off-by: Randy Dunlap <rdunlap@xenotime.net>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Here is the first big update chunk of sound stuff for 3.4-rc1.
In the common sound infrastructure, there are a few changes for
dynamic PCM support (used in ASoC) and a few clean-ups. Majority of
changes are found, as usual, in HD-audio and ASoC.
Some highlights of HD-audio changes:
- All the long-standing static quirk codes for Realtek codec were
finally removed by fixing and extending the Realtek auto-parser.
- The mute-LED control is standardized over all HD-audio codec
drivers using the extended vmaster hook.
- The vmaster slave mixer elements are initialized to 0dB as default
so that the user won't be annoyed by the silent output after
updates, e.g. due to the additions of new elements.
- Other many fix-ups for the misc HD-audio devices.
In the ASoC side, this is a very active release, including a quite a
few framework enhancements. Some highlights:
- Support for widgets not associated with a CODEC, an important part
of the dynamic PCM framework.
- A library factoring out the common code shared by dmaengine based
DMA drivers contributed by Lars-Peter Clausen. This will save a lot
of code and make it much easier to deploy enhancements to
dmaengine.
- Support for binary controls, used for providing runtime
configuration of algorithm coefficients.
- A new DAPM widget type for regulator supplies allowing drivers for
devices that can power down unused supplies while active to do
without any per-driver code.
- DAPM widgets for DAIs, initially giving a speed boost for playback
startup and shutdown and also the basis for CODEC<->CODEC DAI link
support.
- Support for specifying the number of significant bits on audio
interfaces, useful for allowing applications to know how much effort
to put into generating data for a larger sample format.
- Conversion of the FSI driver used on some SH processors to
DMAEngine.
- Conversion of EP93xx drivers to DMAEngine.
- New CODEC drivers for Maxim MAX9768 and Wolfson Microelectronics
WM2200.
- Move audmux driver from arc/arm to sound/soc
- McBSP move from arch/ to sound/ and updates
Also, a few small updates and fixes for other drivers like au88x0,
ymfpci, USB 6fire, USB usx2yaudio are included.
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull updates of sound stuff from Takashi Iwai:
"Here is the first big update chunk of sound stuff for 3.4-rc1.
In the common sound infrastructure, there are a few changes for
dynamic PCM support (used in ASoC) and a few clean-ups. Majority of
changes are found, as usual, in HD-audio and ASoC.
Some highlights of HD-audio changes:
- All the long-standing static quirk codes for Realtek codec were
finally removed by fixing and extending the Realtek auto-parser.
- The mute-LED control is standardized over all HD-audio codec
drivers using the extended vmaster hook.
- The vmaster slave mixer elements are initialized to 0dB as default
so that the user won't be annoyed by the silent output after
updates, e.g. due to the additions of new elements.
- Other many fix-ups for the misc HD-audio devices.
In the ASoC side, this is a very active release, including a quite a
few framework enhancements. Some highlights:
- Support for widgets not associated with a CODEC, an important part
of the dynamic PCM framework.
- A library factoring out the common code shared by dmaengine based
DMA drivers contributed by Lars-Peter Clausen. This will save a
lot of code and make it much easier to deploy enhancements to
dmaengine.
- Support for binary controls, used for providing runtime
configuration of algorithm coefficients.
- A new DAPM widget type for regulator supplies allowing drivers for
devices that can power down unused supplies while active to do
without any per-driver code.
- DAPM widgets for DAIs, initially giving a speed boost for playback
startup and shutdown and also the basis for CODEC<->CODEC DAI link
support.
- Support for specifying the number of significant bits on audio
interfaces, useful for allowing applications to know how much
effort to put into generating data for a larger sample format.
- Conversion of the FSI driver used on some SH processors to
DMAEngine.
- Conversion of EP93xx drivers to DMAEngine.
- New CODEC drivers for Maxim MAX9768 and Wolfson Microelectronics
WM2200.
- Move audmux driver from arc/arm to sound/soc
- McBSP move from arch/ to sound/ and updates
Also, a few small updates and fixes for other drivers like au88x0,
ymfpci, USB 6fire, USB usx2yaudio are included."
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (446 commits)
ASoC: wm8994: Provide VMID mode control and fix default sequence
ASoC: wm8994: Add missing break in resume
ASoC: wm_hubs: Don't actively manage LINEOUT_VMID_BUF
ASoC: pxa-ssp: atomically set stream active masks
ASoC: fsl: p1022ds: tell the WM8776 codec driver that it's the master
ASoC: Samsung: Added to support mono recording
ALSA: hda - Fix build with CONFIG_PM=n
ALSA: au88x0 - Avoid possible Oops at unbinding
ALSA: usb-audio - Fix build error by consitification of rate list
ASoC: core: Fix obscure leak of runtime array
ALSA: pcm - Avoid GFP_ATOMIC in snd_pcm_link()
ALSA: pcm: Constify the list in snd_pcm_hw_constraint_list
ASoC: wm8996: Add 44.1kHz support
ALSA: hda - Fix build of patch_sigmatel.c without CONFIG_SND_HDA_POWER_SAVE
ASoC: mx27vis-aic32x4: Convert it to platform driver
ALSA: hda - fix printing of high HDMI sample rates
ALSA: ymfpci - Fix legacy registers on S3/S4 resume
ALSA: control - Fixe a trailing white space error
ALSA: hda - Add expose_enum_ctl flag to snd_hda_add_vmaster_hook()
ALSA: hda - Add "Mute-LED Mode" enum control
...
This patch adds a new position_fix option value, 4, as a combo mode
to use LPIB for playbacks and POSBUF for captures. It's the way
recommended by Intel hardware guys.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Resitance is futile. The remaining static model quirks for Apple
machines with ALC882-compatible codecs are converted to the auto-parser
now. We can remove all alc*_quirks.c finally.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Finally the all static quirks for ALC880 are converted to the
auto-parser. Since we are never sure whether the BIOS on so many old
machines are really correct, the quirk table entries are copied as
they are, but just providing the proper pin-config values
accordingly.
Since alc880_quirks.c is removed, alc882_quirks.c has to be adjusted
slightly to be built again. There might be some compile warnings due
to the remaining alc882 quirks, but these shall be killed sooner or
later, I don't care it much at this point.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It turned out that BIOS on most of ASUS mobo's set the pin-config tables
reasonably well for the auto-parser. We'd need GPIO setups, but should
work as is other than that.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASUS Z71V has a totally broken BIOS setup (at least the info I got),
thus we need to override the whole pin-config table to make the
auto-parser working correctly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The model=uniwill would work almost as is, but a couple of adjustments
are needed to make the mutli-io working correctly. The headphone and
speaker pins have to be marked properly in pin configs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Similar as the previous patch for model=fujitsu, we can now move the
static quirk for F1734 to the auto-parser. The only difference is the
default pin configurations: F1734 has less pins than Amilo's.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now adding the support for the volume-knob widget, we can move the static
quirk for ALC880 model=fujitsu to the auto-parser completely.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clevo machines with ALC880 are all well with proper BIOS setup.
It seems still requiring the additional COEF setup for the EAPD.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Medion W810 with ALC880 has a typical BIOS bug, copying the
pin-defaults without disabling the unused pins. At least, the pin
0x17 must be disabled. Also, it requires GPIO-2 setup.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC880 model=lg could work fine with the auto-parser due to the recent
rewrite, but it still needs the manual adjustment; namely, the BIOS leaves
unused pins as some real active jacks. This confuses the parser.
Thus we just cover these pins and override the pin-configs as a fix-up.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now we can clean up all static quirks for ALC260.
Also many codes in alc_quirks.c can be ripped off since they have been
used only by ALC260 static quirks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The support for Replacer 627V in the auto-parser needs the unique unsol
event handling: although the machine has a single output pin 0x0f, it's
used for both the headphone and the speaker, and the driver needs to
toggle the output route via GPIO 1.
In addition, it needs a special COEF setup with 0x3050.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ALC260 model=acer needs GPIO1 setup. It could be selected well
if the codec SSID is set properly by BIOS, but to make sure, enable it
forcibly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The model=will for ALC260 requires the pin 0x0f to be a headphone and
some special verbs for the COEF to turn on the amp. Now added these as
fixup entries and removed the static model quirk.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit b99a776d0b removed all effects of
the STAC92HD83* model quirk "hp". However, it left the model selection
and documentation behind, confusing users with inverted mute
leds. Completely remove this quirk and its documentation.
Signed-off-by: Gustavo Maciel Dias Vieira <gustavo@sagui.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the not working internal mic on Dell Vostro 3500 laptop by introducing the
new model dell-vostro-3500.
Signed-off-by: Julian Wollrath <jwollrath@web.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that model=ultra is supported well by the auto-parser, we can get rid
of the whole alc262_quirks.c and its related codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Channel Mode
This is an enum control to change the surround-channel setup,
appears only when the surround channels are available.
It gives the number of channels to be used, "2ch", "4ch" abd "6ch".
According to the configuration, this also controls the
jack-retasking of multi-I/O jacks.
* Independent HP
When this enum control is enabled, the headphone output is routed
from an individual stream (the third PCM such as hw:0,2) instead of
the primary stream.
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the ugly real world, there area really broken devices that don't set
codec SSID correctly. In such a case, the ID can be random, thus the
patching won't work reliably.
For applying the patch forcibly to such a device, the driver will skip
the vendor and/or subsystem ID checks when zero or a negative number is
given in [codec] section.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a new option "snoop" for the traffic control of the HD-audio
controller chip. When set to 0, the non-snooping mode is used with
the traffic control bit is set in each stream control register.
This may allow better operations in the low power mode, but the actual
implementation is depending pretty much on the chipset.
As already implemented, more or less each chipset has own snoop-control
register bit. Now this setup refers to the snoop option, too.
Also, a new VIA chipset may require the non-snooping mode when set so
in BIOS. In such a case, the option value is overridden.
As default, it's still set to snoop=1 for keeping the same behavior as
before. In near future, it'll be set to 0 as default after checking
it works in every system well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce the pincfg table to patch_conexant.c for fixing up the extra
pin-configuration for auto-parser. As an example, Lenovo X200 model is
replaced with this new mechanism. (This also fixes the wrong mixer
elements for docking-station I/O in the previous model quirk
automagically.)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Similarly like ALC662 asus-mode* models, rewrite the laptop-amic and
dmic models with the static pin-config tables.
Now we can get rid of all alc269_quirks.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Let's remove the rest of ALC861 and ALC861-VD quirks.
If any breakage is found, it can be fixed easily via the pin-config
table update.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Implement new fixup entries for Quanta FL1 and Fujitsu Lifebook
specific COEF and pin configurations. Removed the model entries
from alc269_quirks.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC260 HP models work with the BIOS auto-parser. Let's cut them off.
Also move alc260_hp_master_*() to alc262_quirks.c as these are still
referred from there.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add new parameter to disable rounding of buffer/period sizes to
multiples of 128 bytes. This is more efficient in terms of memory
access but isn't required by the HDA spec and prevents users from
specifying exact period/buffer sizes. For example for 44.1kHz, a
period size set to 20ms will be rounded to 19.59ms.
Tested and enabled on Intel HDA controllers. Option is disabled by
default for other controllers.
Tested-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a new driver for supporting Digigram Lola PCI-e boards.
Lola has a similar h/w design like HD-audio but with extended verbs.
Thus the driver is written similarly like HD-audio driver in the bus
part. The codec part is rather written in a fixed way specific to the
Lola board because of the verb incompatibility.
The driver provides basic PCM, supporting multi-streams and mixing.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
alc662 series only have 3 DAC, so it can only support 5stack-dig
instead of 6stack-dig.
[updated HD-Audio-Models.txt as well by tiwai]
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix some minor typos:
* informations => information
* there own => their own
* these => this
Signed-off-by: Sylvestre Ledru <sylvestre.ledru@scilab.org>
Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
BugLink: http://bugs.launchpad.net/bugs/701271
This new model, named "asus", is identical to the "hp_laptop" model,
except for the location of the internal mic, which is at pin 0x1a.
It is used for Asus K52JU and Lenovo G560.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch is only for RFC purpose of ASoC documentation updates which
match with current ASoC codes with documents. Mostly modify features
are modified to be sync with changes after multi-component patches.
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add experimental support for the Asus Xonar HDAV1.3 Slim sound card.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add experimental support for the Asus Xonar DG sound card.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for the AuzenTech X-Meridian 7.1 2G sound card.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for the TempoTec/MediaTek HiFier Serenade sound card.
The PCI ID was already there, but the driver handled it like the
Fantasia model, which resulted in a dummy recording device. As
a stereo output-only card, this model is to be handled exactly
like the HG2PCI.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for the Kuroutoshikou CMI8787-HG2PCI sound card.
[replaced non-latin letters in the patch by tiwai]
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The snd-hifier driver contains more duplicated code than model-specific
code, so it does not make sense for it to be a separate driver.
Handling the two-channel output restriction can be easily done in the
generic driver.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
What was previously known as via_dmapos_patch, and hard-coded to be
used for VIA and ATI controllers, is now configurable through a module
option. The background is that some VIA controllers seem to prefer
via_dmapos_patch to be turned off.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This new model adds the following functionality to HP G60:
- Automute of internal speakers
- Autoswitch of internal/external mics
- Remove SPDIF not physically present
BugLink: http://launchpad.net/bugs/587388
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Its hardware is handled more fully by the new azt1605/azt2316 drivers.
Signed-off-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a new driver for Aztech Sound Galaxy ISA soundcards based on the
AZT1605 and AZT2316 chipsets. It's constructed as two seperate drivers
for either chipset generated from the same source file, with (very)
minimal ifdeffery.
The drivers check the SB DSP version to decide if they are being loaded
for the right chip. AZT1605 returns 2.1 by default and AZT2316 3.1.
This isn't full-proof as the DSP version can actually be set through
software but it's close enough -- as far as I've been able to see, the
DSP version can not be stored in the EEPROM and the cards will therefore
startup with the defaults.
This distinction could (with the same success rate) also be used to
decide which chip we're looking at at runtime meaning a single, merged
driver is also an option but I feel it's actually nicer this way. A
merged driver would have to postpone translating the passed in resource
values to the card configuration until it knew which one it was looking
at and would need to postpone erring out on mpu_irq=10 for azt1605 and
mpu_irq=3 for azt2316.
The drivers have been tested on various cards. For snd-azt1605:
FCC-ID I38-MMSN811: Aztech Sound Galaxy Nova 16 Extra
FCC-ID I38-MMSN822: Aztech Sound Galaxy Pro 16 II
and for snd-azt2316:
FCC-ID I38-MMSN824: Aztech Sound Galaxy Pro 16 AB
FCC-ID I38-MMSN826: Trust Sound Expert DeLuxe Wave 32 (05201)
FCC-ID I38-MMSN830: Trust Sound Expert DeLuxe 16+ (05202)
FCC-ID I38-MMSN837: Packard Bell ISA Soundcard 030069
FCC-ID I38-MMSN846: Trust Sound Expert DeLuxe 16-3D (06300)
FCC-ID I38-MMSN847: Trust Sound Expert DeLuxe Wave 32-3D (06301)
FCC-ID I38-MMSN852: Aztech Sound Galaxy Waverider Pro 32-3D
826 and 846 were also marketed directly by Aztech and then known as:
FCC-ID I38-MMSN826: Aztech Sound Galaxy Waverider 32+
FCC-ID I38-MMSN846: Aztech Sound Galaxy Nova 16 Extra II-3D
Together, these cover the AZT1605 and AT2316A, AZT2316R and AZT2316-S
chipsets. All cards work fully -- full-duplex PCM, MIDI and FM. Full
duplex is a little flaky on some.
I38-MSN811 tends to not work in full-duplex but sometimes does with the
highest success rate being achieved when you first start the capture and
then a playback instead of the other way around (it's a CS4231-KL
codec).
The cards with an AD1845XP codec (my I38-MMSN826 and one of my
I38-MMSN830s) are also somewhat duplex-challenged. Sometimes full-duplex
works, sometimes not and this varies from try to try. This seems likely
to be a timing problem somewhere inside wss-lib.
I38-MMSN826 has an additional "ICS2115 WaveFront" wavetable synth
onboard that isn't supported yet. The wavetable synths on I38-MMSN847
and I38-MMSN852 are wired directly to the standard MPU-401 UART and the
AUX1 input on the codec and work without problem.
CD-ROM audio on the cards is routed to the codec "Line" input, Line-In
to its Aux input, and FM/Wavetable to its AUX1 input. I did not rename
the controls due to the capture source enumeration: I see that
capture-source overrides are hardcoded in wss-lib and this is just too
ugly to live.
Versus the old snd-sgalaxy driver these drivers add support for the
models without a configuration EEPROM (which are common), full-duplex,
MPU-401 UART and OPL3. In the future they might grow support for that
ICS2115 WaveFront synth on 826 and an hwdep interface to write to the
EEPROM on the models that have one.
Signed-off-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
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
Add documentation about the autodetection of the VIA codec models to
avoid the false impression that they are not supported.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai