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.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v2.0.18 (GNU/Linux)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=+JSm
-----END PGP SIGNATURE-----
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>
Merge with latest Linus' tree, as I have incoming patches
that fix code that is newer than current HEAD of for-next.
Conflicts:
drivers/net/ethernet/realtek/r8169.c
Install commands should not be used to specify soft dependencies among
modules. When loading modules it's much better to have a softdep that
modprobe knows what's being done than having to fork/exec another
instance of modprobe to load the other module.
By using a softdep user has also an option to remove the dependencies
when removing the module (and if its refcount dropped to 0)
Signed-off-by: Lucas De Marchi <lucas.demarchi@profusion.mobi>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
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>
This patch updates Jonathan Woithe's contact details across the kernel tree.
Signed-off-by: Jonathan Woithe <jwoithe@just42.net>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
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.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v2.0.18 (GNU/Linux)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=5FJq
-----END PGP SIGNATURE-----
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>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (549 commits)
ALSA: hda - Fix ADC input-amp handling for Cx20549 codec
ALSA: hda - Keep EAPD turned on for old Conexant chips
ALSA: hda/realtek - Fix missing volume controls with ALC260
ASoC: wm8940: Properly set codec->dapm.bias_level
ALSA: hda - Fix pin-config for ASUS W90V
ALSA: hda - Fix surround/CLFE headphone and speaker pins order
ALSA: hda - Fix typo
ALSA: Update the sound git tree URL
ALSA: HDA: Add new revision for ALC662
ASoC: max98095: Convert codec->hw_write to snd_soc_write
ASoC: keep pointer to resource so it can be freed
ASoC: sgtl5000: Fix wrong mask in some snd_soc_update_bits calls
ASoC: wm8996: Fix wrong mask for setting WM8996_AIF_CLOCKING_2
ASoC: da7210: Add support for line out and DAC
ASoC: da7210: Add support for DAPM
ALSA: hda/realtek - Fix DAC assignments of multiple speakers
ASoC: Use SGTL5000_LINREG_VDDD_MASK instead of hardcoded mask value
ASoC: Set sgtl5000->ldo in ldo_regulator_register
ASoC: wm8996: Use SND_SOC_DAPM_AIF_OUT for AIF2 Capture
ASoC: wm8994: Use SND_SOC_DAPM_AIF_OUT for AIF3 Capture
...
* 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>
There are numerous broken references to Documentation files (in other
Documentation files, in comments, etc.). These broken references are
caused by typo's in the references, and by renames or removals of the
Documentation files. Some broken references are simply odd.
Fix these broken references, sometimes by dropping the irrelevant text
they were part of.
Signed-off-by: Paul Bolle <pebolle@tiscali.nl>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
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>
* '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
...
Below you will find an updated version from the original series bunching all patches into one big patch
updating broken web addresses that are located in Documentation/*
Some of the addresses date as far far back as 1995 etc... so searching became a bit difficult,
the best way to deal with these is to use web.archive.org to locate these addresses that are outdated.
Now there are also some addresses pointing to .spec files some are located, but some(after searching
on the companies site)where still no where to be found. In this case I just changed the address
to the company site this way the users can contact the company and they can locate them for the users.
Signed-off-by: Justin P. Mattock <justinmattock@gmail.com>
Signed-off-by: Thomas Weber <weber@corscience.de>
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Cc: Paulo Marques <pmarques@grupopie.com>
Cc: Randy Dunlap <rdunlap@xenotime.net>
Cc: Michael Neuling <mikey@neuling.org>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
Lenovo IdeaPad Y430 has an additional subwoofer connected at pin 0x1b,
which isn't muted when headphone is plugged in. This adds additional
support to the extra subwoofer via new ideapad model.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move sound (OSS & ALSA) kernel parameters to their own files.
Signed-off-by: Randy Dunlap <randy.dunlap@oracle.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: (250 commits)
ALSA: hda: Storage class should be before const qualifier
ASoC: tpa6130a2: Remove CPVSS and HPVdd supplies
ASoC: tpa6130a2: Define output pins with SND_SOC_DAPM_OUTPUT
ASoC: sdp4430 - add sdp4430 pcm ops to DAI.
ASoC: TWL6040: Enable earphone path in codec
ASoC: SDP4430: Add support for Earphone speaker
ASoC: SDP4430: Add sdp4430 machine driver
ASoC: tlv320dac33: Avoid powering off while in BIAS_OFF
ASoC: tlv320dac33: Use dev_dbg in dac33_hard_power function
ALSA: sound/pci/asihpi: Use kzalloc
ALSA: hdmi - dont fail on extra nodes
ALSA: intelhdmi - add id for the CougarPoint chipset
ALSA: intelhdmi - user friendly codec name
ALSA: intelhdmi - add dependency on SND_DYNAMIC_MINORS
ALSA: asihpi: incorrect range check
ALSA: asihpi: testing the wrong variable
ALSA: es1688: add pedantic range checks
ARM: McBSP: Add support for omap4 in McBSP driver
ARM: McBSP: Fix request for irq in OMAP4
OMAP: McBSP: Add 32-bit mode support
...
The ESS ES968 chip is nothing more then a PnP companion
for a non-PnP audio chip. It was paired with non-PnP ESS' chips:
ES688 and ES1688. The ESS' audio chips are handled by the es1688
driver in native mode. The PnP cards are handled by the ES968
driver in SB compatible mode.
Move the ES968 chip handling to the es1688 driver so the driver
can handle both PnP and non-PnP cards. The es968 is removed.
Also, a new PnP id is added for the card I acquired (the change
was tested on this card).
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix obvious cases of "it's" being used when "its" was meant.
Signed-off-by: Francis Galiegue <fgaliegue@gmail.com>
Acked-by: Randy Dunlap <rdunlap@xenotime.net>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
Added the support of AudioScience ASI boards.
The driver has been tested for years on alsa-driver external tree,
now finally got merged to the kernel.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The probe_only module parameter skips the codec initialization, too.
Remove the model=hwio code and use second bit in probe_only to
skip the HDA codec reset procedure.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Using the 'model=hwio' option, the driver bypasses any codec
initialization and the reset procedure for codecs is also
bypassed. This mode is usefull to enable direct access using
hwdep interface (using hdaverb or hda-analyzer tools) and
retain codec setup from BIOS.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Add support for the Edirol UA-1000 to the UA-101 driver.
Both devices behave the same, so we just have to shuffle around some
interface numbers and name strings.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
BugLink: https://bugs.edge.launchpad.net/ubuntu/+source/linux/+bug/343989
Add a model quirk for the NVIDIA based Macmini hardware, aka Macmini 3,1. The
pinout is almost identical to the mb5 quirk, except for no microphone and
the line-in mixer controls being on a different index. Everything works in
2ch mode, but as I am not sure what needs to be changed for 6ch mode, or
whether the Mac Mini's chip supports 6ch mode, I have simply duplicated
the code from the mb5 quirk for the mac mini chmode management. The new
model parameter for this quirk is "macmini3".
Signed-off-by: Luke Yelavich <luke.yelavich@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add patch for the Conexant 5066 HDA codec to support the Lenovo IdeaPad U150
Signed-off-by: Greg Alexander <greigs@galexander.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow the override of vendor-id, subsystem-id, revision-id and chip name
via patch loading. Updated the document, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the support for Toshiba Satellite M300 with Conexant 5051 codec.
Since the laptop has no port C connection and the pin reports always
the jack sense true, we need to ignore port-C unsol event.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Minor fixes for HP Compaq Presario F700 quirks with Cxt5051 codec:
- changed the capture mixer elements to the standard name.
- fixed the quirk name string without a space
- sorted the quirk list
- updated the documentation
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Narrow the dma and irq selection after the DOS driver.
Add ALSA configuration description as well.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With the attached patch I am able to use the sound on a new IMac 27.
What works:
*) Internal speakers
*) Internal microphone
*) Headphone
I don't have an external mic or a SPDIF device to test the rest.
Signed-off-by: Rafael Avila de Espindola <rafael.espindola@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add experimental support for the Edirol UA-101 audio/MIDI interface.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just to save some time, add direct git link to grub hda-analyzer
Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is an updated patch for the Apple iMac 9,1 model to add sound.
Original patch posted here:
http://article.gmane.org/gmane.linux.alsa.devel/61361/match=
I have been using this patch for a while now
and have to say it works vary well, except for a few minor
things:
With the iMac 24-inch 3.06GHz Intel Core 2 Duo
everything seems to be working as it should,
although I have not looked into the microphone
(never really use one, nor have any apps to test,
my guess is it doesn't work, or I never figured out how
to get it to work).
With the iMac 24-inch 2.66GHz Intel Core 2 Duo
everything is the same as with the above machine
except I'm hearing a light scratchy/distortion noise
come out of the speakers when using headphones(above machine
does not do this).
Other than that the sound level is great(especially with good Dj headphones).
Signed-off-by: Justin P. Mattock <justinmattock@gmail.com>
Tested-by: Justin P. Mattock <justinmattock@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The mute-LED isn't synchronized with the actual mute state on some
HP laptops with IDT 92HD83xxx codecs. A similar hack using
check_power_status callback is added for this codec, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To unify control names, rename "PC Speaker" to "Speaker" for PPC ALSA drivers.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To avoid confusion in control names for the standard analog PC Beep generator
using a small Internal PC Speaker, rename all related "PC Speaker" and "PC
Beep" controls to "Beep" only. This name is more universal and can be also
used on more platforms without confusion.
Introduce also "Internal Speaker" in ControlNames.txt for systems with
full-featured build-in internal speaker.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, if the high-res timers are unavailable, snd-pcsp does not
initialize. People who choose it over pcspkr, loose their console beeps
in that case and get annoyed.
With this patch, the console beeps remain regardless of the high-res
timers. Additionally, the "nopcm" modparam is added to forcibly
disable the PCM capabilities of the driver.
Signed-off-by: Stas Sergeev <stsp@aknet.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On FSC laptops, the sound gets muted gradually when the volume is chnaged.
This is due to the wrong volume-knob widget setup. The delta bit (bit 7)
shouldn't be set for these devices.
This patch adds a new quirk to set the value 0x7f to the widget 0x24
instead of 0xff.
Reference: Novell bnc#546006
http://bugzilla.novell.com/show_bug.cgi?id=546006
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Old Soundscape cards (pre PnP) work only with AD1848 codecs.
If the CS4231 codec is installed it must be used in AD1848
compatible mode.
Also, add gameport support and remove an unused mpu field.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The conversion solves the problem that firmware size was set to 64KB
while non PnP cards have 128KB firmware files.
An additional firmware initialization code has been moved from the OSS
driver.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reference: ALSA bug #0004614https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4614
port-A (0x11) - front hp-out
port-D (0x12) - rear line out
port-E (0x1c) - front mic-in
port-F (0x16) - Internal speakers
digital-mic (0x17) - Internal mic
init verbs, mixers, jack sensing and PCI_QUIRK to support this hardware
Signed-off-by: Miguel de Barros <miguel.de.barros@bluewin.ch>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* topic/hda: (92 commits)
ALSA: hda - Use auto model for HP laptops with ALC268 codec
ALSA: hda/realtek: Added support for CLEVO M540R subsystem, 6 channel + digital
ALSA: hda - Add support of Alienware M17x laptop
ALSA: hda - Remove dead codes from patch_sigmatel.c
ALSA: hda - Fix input source selection of IDT92HD73xx
ALSA: hda - Fix obsolete CONFIG_SND_DEBUG_DETECT
ALSA: hda - Unmute docking line-out as default with AD1984A codec
ALSA: hda - Add another entry for Nvidia HDMI device
ALSA: hda - Add missing GPIO initialization for AD1984A laptop model
ALSA: hda - Add support of docking auto-mute/mic for AD1984A laptop model
ALSA: hda - Fix ALC268/ALC269 headphone pin routing
ALSA: hda - Create "Digital Mic Capture Volume" correctly for IDT codecs
ALSA: hda - Add more quirk for HP laptops with AD1984A
ALSA: hda - Add / fix model entries for HD-audio driver
ALSA: hda - Add full audio support on Acer Aspire 7730G notebook
ALSA: hda - Improve auto-cfg mixer name for ALC662
ALSA: hda - Improve auto-cfg mixer name for ALC861-VD
ALSA: hda - Improve auto-cfg mixer name for ALC262
ALSA: hda - Improve auto-cfg mixer name for ALC260
ALSA: hda - Improve auto-cfg mixer name for ALC880
...
Instead of allocating the real buffers, use a fake buffer and ignore
read/write in the dummy driver so that we can save the resources.
For mmap, a single page (unique to the direction, though) is reused
to all buffers.
When the app requires to read/write the real buffers, pass fake_buffer=0
module option at loading time. This will get back to the old behavior.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The model clevo-m540r was created with 6-channel and digital support. All
functions verified except spdif. Tested with a VIT D2000 laptop which has:
[lspci extract]
Audio device [0403]: Intel Corporation 82801H (ICH8 Family) HD Audio
Controller [8086:284b] (rev 03)
Subsystem: CLEVO/KAPOK Computer Device [1558:5409]
[/proc/asound/card0/codec\#0 header]
Codec: Realtek ALC883
Address: 0
Function Id: 0x1
Vendor Id: 0x10ec0883
Subsystem Id: 0x15585409
Revision Id: 0x100002
[Added a comment about HP mute and the model description by tiwai]
Signed-off-by: Dhionel Diaz <ddiaz@cenditel.gob.ve>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the quirk for Alienware M17x with IDT 92HD73* codec chip.
It has two HP and one line-out jack, one mic jack, a built-in
speaker and a built-in mic.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add debug module option to snd core.
This controls the debug print level. When CONFIG_SND_DEBUG_VERBOSE
is set, you can suppress the debug messages by giving or changing this
parameter to a lower value. debug=0 means no debug messsages.
As default, it's set to the verbose level 2.
Since this option can be changed dynamically via sysfs file, you can
suppress the verbose debug messages on the fly, which wasn't possible
before.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The sentense "Unknown model for xxx, ..." makes people too nervous
and drives them to a direction to a wrong "fix" by giving any
mismatching model option.
Let's rephrase the messages to be more nice and easy (at least that
won't make people suspect conspiracies).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the logging functionality to xrun_debug to record the hwptr
updates via snd_pcm_update_hw_ptr() and snd_pcm_update_hwptr_interrupt(),
corresponding to 16 and 8, respectively.
For example,
# echo 9 > /proc/asound/card0/pcm0p/xrun_debug
will record the position and other parameters at each period interrupt
together with the normal XRUN debugging.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge patch_alc882() and patch_alc883() to the former one since both
codecs have fairly similar connections but just a slight difference.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Samsung P50 requires the HP auto-muting unlike other Samsung models.
Added a new model=samsung-p50 to support this.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* topic/oxygen:
sound: virtuoso: add Xonar Essence ST support
sound: virtuoso: enable HDAV S/PDIF input
sound: virtuoso: add another DX PCI ID
sound: oxygen: reset DMA when stream is closed
* topic/maya44:
ALSA: ice1724 - Add ESI Maya44 support
ALSA: ice1724 - Allow spec driver to create own routing controls
ALSA: ice1724 - Add PCI postint to reset sequence
ALSA: ice1724 - Clean up definitions of DMA records
ALSA: ice1724 - Check error in set_rate function
* topic/asoc: (135 commits)
ASoC: Apostrophe patrol
ASoC: codec tlv320aic23 fix bogus divide by 0 message
ASoC: fix NULL pointer dereference in soc_suspend()
ASoC: Fix build error in twl4030.c
ASoC: SSM2602: assign last substream to the master when shutting down
ASoC: Blackfin: document how anomaly 05000250 is handled
ASoC: Blackfin: set the transfer size according the ac97_frame size
ASoC: SSM2602: remove unsupported sample rates
ASoC: TWL4030: Check the interface format for 4 channel mode
ASoC: TWL4030: Use reg_cache in twl4030_init_chip
ASoC: Initialise dev for the dummy S/PDIF DAI
ASoC: Add dummy S/PDIF codec support
ASoC: correct print specifiers for unsigneds
ASoC: Modify mpc5200 AC97 driver to use V9 of spin_event_timeout()
ASoC: Switch FSL SSI DAI over to symmetric_rates
ASoC: Mark MPC5200 AC97 as BROKEN until PowerPC merge issues are resolved
ASoC: Fabric bindings for STAC9766 on the Efika
ASoC: Support for AC97 on Phytec pmc030 base board.
ASoC: AC97 driver for mpc5200
ASoC: Main rewite of the mpc5200 audio DMA code
...
Added 7.1 support for MSI GX620 and jack quirk.
Reference: kernel bug#13430
http://bugzilla.kernel.org/show_bug.cgi?id=13430
Signed-off-by: David Heidelberger <d.okias@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
with BIOS probing only we offer a non functional headphone swith and
volume slider.
Signed-off-by: Guido Günther <agx@sigxcpu.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Short story: this laptop has 5.1 built-in speakers which you *really*
want to use (the not-so-"sub" woofer is what makes the audio above
average for a laptop), so 6-channel support is important (plus a decent
asound.conf to upmix stereo). It also has the 3 typical jacks that ought
to have a selectable mode. And it's based on ALC889, which sucks.
Rationale/explanations:
The const_channel_count stuff was added because, for a laptop like this,
you always have 6 channels available (internal speakers) but still need
to set the mode for the 3 external jacks. Therefore, the device always
needs to be in 6-channel mode but there still needs to be a mixer
control for the jack mode. You could use line/mic-in at the same time as
the 6 internal speakers, for example. You might be tempted to make it
even smarter by dynamically switching the max channel count when
headphones are plugged in (therefore muting the internal speakers and
reducing the physical channel count to the jack channel mode), but as a
user I consider this to be harmful because I want the audio to blow up
to 6 channels / upmixed as soon as I unplug the headphones, and having
opened the device while in 2-channel mode would prevent this from
working (and always making 6-channel mode available doesn't do any harm).
The hardware needs EAPD turned on and the DACs routed to the internal
speaker pins, so the patch adds those verbs.
The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT work
by default, at least here. I wasted much time trying to talk to
Realtek/pshou about this, but they just kept sending me useless updates
to patch_realtek.c that did nothing relevant. In the end I gave up and
brute forced the issue by trying to flip every bit in the proprietary
coefficient registers, and eventually found the two magic registers that
need to be cleared to enable all DACs. I have only heard Acer users
complain, but that might be because ALC889 is pretty new and using 5.1
(and noticing the missing center/lfe channels) might not be that common.
If this is a generalized issue with all ALC889 systems then those verbs
should probably be moved to a common verb array.
The internal mic is untested and probably doesn't work.
These settings will probably work for other Acer Gemstone laptops with
the same 5.1 speaker config. When identified, those should be added to
the PCI subsystem ID list.
Signed-off-by: Hector Martin <hector@marcansoft.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM hw_ptr jiffies check results sometimes in problems when a
hardware doesn't give smooth hw_ptr updates. So far, au88x0 and some
other drivers appear not working due to this strict check.
However, this check is a nice debug tool, and the capability should be
still kept.
Hence, we disable this check now as default unless the user enables it
by setting the xrun_debug mode to the specific stream via a proc file.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for the Asus Xonar Essence ST and its daughterboard.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent fix for the headphone volume control on IDT/STAC codecs
resulted in the removal of invalid "Side" volume eventually. But,
if the front panel doesn't exist, this setup could be regarded as a
sort of regression, as reported in kernel bug #13250.
Now as a workaround, a new model 5stack-no-fp is added so that the user
without the front panel can choose this one explicitly.
Reference: bko#13250
http://bugzilla.kernel.org/show_bug.cgi?id=13250
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support for MacBook 3,1 sound by adding a model new
"mb31" with the appropriate init verbs, mixers and channel modes to
the ALC883 configuration. patch_alc882() and patch_alc883() are
modified to handle the MacBook 3,1 sound-chip (Realtek ALC889A)
correctly.
Signed-off-by: Torben Schulz <public@letorbi.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add specific configuration for Samsung NC10 mini notebook. Internal
mic/speakers will be correctly muted when plugging in external ones.
Mixer controls are added for speakers, headphones and PC beep.
"Boost" is added for the microphones.
Signed-off-by: Chris Pockelé <chris.pockele.f1@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the missing descriptions and the model names for Realtek codecs
to the documentation and the config table.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add module parameter to enable or disable
joystick port (gameport) on the SC6600 and
later cards.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Many modern CODECs have shared resources on chip which must be enabled
for portions of the chip to work but which can be disabled at other times
in order to achieve power savings. Examples of such resources include
power supplies and some internal clocks.
Since these widgets are dependencies for the audio path but do not carry
audio signals they require slightly different handling to most widgets -
they do not contribute to the audio path and so should not be counted as
either inputs or outputs during path walks.
Cases where one supply provides a supply for another will require
additional work. There is also room for more optimisation of the graph
walking to avoid repeated checks for the same thing.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move ALSA docbooks to be with the rest of the kernel docbooks and add
them to the Makefile so that they build. Latter required a few minor
changes to alsa .tmpl files.
(I did not remove all of the trailing whitespace in the .tmpl files.)
Fixes kernel bugzilla #12726: http://bugzilla.kernel.org/show_bug.cgi?id=12726
Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Cc: documentation_man-pages@kernel-bugs.osdl.org
Cc: Nicola Soranzo <nsoranzo@tiscali.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the mic input of HP dv6736 with Conexant 5051 codec chip.
This laptop seems have no mic-switching per jack connection.
A new model hp-dv6736 is introduced to match with the h/w implementation.
Reference: Novell bnc#480753
https://bugzilla.novell.com/show_bug.cgi?id=480753
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the model=auto to STAC/IDT codecs to use the BIOS default setup
explicitly. It can be used to disable the device-specific model quirk
in the driver.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make user_pin overriding even the driver pincfg, e.g. the static / fixed
pin config table in patch_sigmatel.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rename from override_pin and cur_pin with user_pin and driver_pin,
respectively, to be a bit more intuitive.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is no OSS cs4232 driver in the kernel
any more and this documentation does not
contain any info useful for ALSA driver.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently the HP connector on X200 dock doesn't detect when a HP is connected
nor allows sound to be played using it. This patch fixes the problem by adding
a quirk for this specific model. It's possible that others have the same NID
(0x19) to report when dock HP is connected, but I don't have access to any.
Please Cc me in the reply since I'm not subscribed to alsa-devel@.
Signed-off-by: Aristeu Rozanski <aris@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As just pointed out to me, the new tyan model for ALC262 was
implemented but not documented. This adds the board to the
list, using both its marketing name (Thunder n6650W) and its
model number (S2915-E).
Signed-off-by: Tony Vroon <tony@linx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>