In the realtek auto-probing mode, the initialization of amp with
some magic COEF or EAPD verbs is applied only when the codec SSID
has valid values to satisfy the realtek's definition.
However, many devices don't provide in that way, thus the device
doesn't work as is.
This patch allows the same initialization code even if the SSID
doesn't pass the bit test. Also, alc_subsystem_id() is changed
just to check and define the type, so that it's called in the
parser, instead of the initializer.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed the compile warning below by initializatin iomidi variable properly.
sound/pci/cmipci.c: In function ‘snd_cmipci_probe’:
sound/pci/cmipci.c:3017: warning: ‘iomidi’ may be used uninitialized in this function
Signed-off-by: Subrata Modak <subrata@linux.vnet.ibm.com>
Cc: linux-pci <linux-pci@atrey.karlin.mff.cuni.cz>
Cc: Balbir Singh <balbir@linux.vnet.ibm.com>
Cc: Sachin P Sant <sachinp@linux.vnet.ibm.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current ad1884a-mobile model has a problem that the speaker output
doesn't work sometimes after boot or power-saving on some HP laptops.
It seems that the verbs accessing to the non-functional widgets cause
this problem.
This patch simplifies the init verbs for mobile model not to touch
unnecessary setups so that it avoids the speaker-mute problem.
Reference: Novell bnc#495668
https://bugzilla.novell.com/show_bug.cgi?id=495668
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Appearently, a big delay ~300ms is required before hw is settled and ready
to transfer samples on some hardware variants. Also, return back
"clocking to 48000Hz" message when something fails.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The current error-recovery scheme for the codec communication errors
doesn't work always well. Especially falling back to the
single-command mode causes the fatal problem on many systems.
In this patch, the problematic verb is re-issued again after the error
(even with polling mode) instead of the single-cmd mode. The
single-cmd mode will be used only when specified via the command
option explicitly, mainly just for testing.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cache quries for PCM and STREAM parameters as well as ampcap and
pincap sharing the hash table. This will reduce the superfluous
access of the same codec verbs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Support ASUS F81Se F5Q P80 U20A U80 U50 UX50 for ALC269
- Support ASUS F70SL UX20 X58LE F50Z N80Vc N81Te N505Tp Vx3V N5051A
for ALC663
- Support DELL ZM1 for ALC272
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't unmute unneeded amps for input mixers of ALC662 & co.
It caused possible recording noises.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the missing definition of max channels for CA0110, which resulted
in an error at opening PCM devices.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
(Re)set function_id only from the value on FG nodes.
The current code overrides the value with the last widget.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the headphone can have no unique DAC, the current code doesn't
check the HP-detection although it should. Put the hp-detection check
before the DAC check to fix this bug.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check the length to copy via strlen() beforehand to avoid the stack
corruption, or use strlcpy() to be safe in HD-audio codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the support for Creative SB X-Fi boards with UAA (HD-audio) mode.
In the HD-audio mode, no multiple streams are supported by just it
behaves like a normal HD-audio device.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Subject says it all. Briefly, use hp_only for another Dell Inspiron 8600.
Reference: Ubuntu #41015 (https://launchpad.net/bugs/41015)
Signed-off-by: Daniel T Chen <seven.steps@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While cleaning up quirks, I noticed that there is a duplicated quirk for
the SSID 0x103c0934. Looking back through the bug reports, I've concluded
that there is only one necessary quirk (hp_mute_led), so this patch
removes the conflicting one.
Reference: Ubuntu #44066 (https://launchpad.net/bugs/44066)
Signed-off-by: Daniel T Chen <seven.steps@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit fa00e046b4
added a new bitfield not adjacent to other
bitfields in the same struct. Moved the new one.
Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the key value generation for get/set amp verbs. The upper bits of
the parameter have to be combined with the verb value to be unique for
each direction/index of amp access.
This fixes the resume problem on some hardwares like Macbook after
the channel mode is changed.
Tested-by: Johannes Berg <johannes@sipsolutions.net>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* topic/memdup_user:
ALSA: sound/pci: use memdup_user()
ALSA: sound/usb: use memdup_user()
ALSA: sound/isa: use memdup_user()
ALSA: sound/core: use memdup_user()
* 'master' of git://git.alsa-project.org/alsa-kernel:
[ALSA] intel8x0: add one retry to the ac97_clock measurement routine
[ALSA] intel8x0: fix wrong conditions in ac97_clock measure routine
[ALSA] intel8x0: do not use zero value from PICB register
[ALSA] intel8x0: an attempt to make ac97_clock measurement more reliable
[ALSA] pcm-midlevel: Add more strict buffer position checks based on jiffies
[ALSA] hda_intel: fix unexpected ring buffer positions
Added the models for quirk bitmask 1734:110x and 1734:113x of
Fujitsu laptops.
This will fix the model detection for Amilo Xa3540.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It seems that on some hardware platforms, the first measurement is wrong.
This patch adds second measurement to this case.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Don't call snd_jack_report at release of sigmatel and conexnat codecs
which results in Oops at unloading the module.
The Oops is triggered by the power-up sequence during the free due to
the pincfg restoration. Since the power-up sequence is involved with
the unsol handling, the jack reporting may be issued during that.
The Oops occurs with this jack reporting because the jack instances
have been already released but the codec doesn't do the proper
book-keeping.
This patch adds the book-keeping of jack instances to avoid the access
to bogus pointers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
prototype of a driver for the digigram lx6464es 64 channel ethersound
interface.
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is the second go through of the old DMA_nBIT_MASK macro,and there're not
so many of them left,so I put them into one patch.I hope this is the last round.
After this the definition of the old DMA_nBIT_MASK macro could be removed.
Signed-off-by: Yang Hongyang <yanghy@cn.fujitsu.com>
Cc: Russell King <rmk@arm.linux.org.uk>
Cc: Tony Lindgren <tony@atomide.com>
Cc: "David S. Miller" <davem@davemloft.net>
Cc: James Bottomley <James.Bottomley@HansenPartnership.com>
Cc: Greg KH <greg@kroah.com>
Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
It seems that the zero value from the PICB (position in current buffer)
register is not reliable. Use jiffies to correct returned value
from the ring buffer pointer callback.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- use monotonic posix clock to measure time
- try to avoid reading zero from PICB (position in current buffer) register
- show also measured samples
- when clock is near 41000 or 44100, use exactly these values
(they appears to be reference clocks for hardware manufacturers)
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
I found two issues with ICH7-M (it should be related to other HDA chipsets
as well):
- the ring buffer position is not reset when stream restarts (after xrun) -
solved by moving azx_stream_reset() call from open() to prepare() callback
and reset posbuf to zero (it might be filled with hw later than position()
callback is called)
- irq_ignore flag should be set also when ring buffer memory area is not
changed in prepare() callback - this patch replaces irq_ignore with
more universal check based on jiffies clock
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (36 commits)
ALSA: hda - Add VREF powerdown sequence for another board
ALSA: oss - volume control for CSWITCH and CROUTE
ALSA: hda - add missing comma in ad1884_slave_vols
sound: usb-audio: allow period sizes less than 1 ms
sound: usb-audio: save data packet interval in audioformat structure
sound: usb-audio: remove check_hw_params_convention()
sound: usb-audio: show sample format width in proc file
ASoC: fsl_dma: Pass the proper device for dma mapping routines
ASoC: Fix null dereference in ak4535_remove()
ALSA: hda - enable SPDIF output for Intel DX58SO board
ALSA: snd-atmel-abdac: increase periods_min to 6 instead of 4
ALSA: snd-atmel-abdac: replace bus_id with dev_name()
ALSA: snd-atmel-ac97c: replace bus_id with dev_name()
ALSA: snd-atmel-ac97c: cleanup registers when removing driver
ALSA: snd-atmel-ac97c: do a proper reset of the external codec
ALSA: snd-atmel-ac97c: enable interrupts to catch events for error reporting
ALSA: snd-atmel-ac97c: set correct size for buffer hardware parameter
ALSA: snd-atmel-ac97c: do not overwrite OCA and ICA when assigning channels
ALSA: snd-atmel-ac97c: remove dead break statements after return in switch case
ALSA: snd-atmel-ac97c: cleanup register definitions
...
Replace all DMA_24BIT_MASK macro with DMA_BIT_MASK(24)
Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Replace all DMA_28BIT_MASK macro with DMA_BIT_MASK(28)
Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Replace all DMA_30BIT_MASK macro with DMA_BIT_MASK(30)
Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Replace all DMA_31BIT_MASK macro with DMA_BIT_MASK(31)
Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Replace all DMA_32BIT_MASK macro with DMA_BIT_MASK(32)
Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Add powerdown sequence for VREF using a shared jack when the headphone
is present and the microphone isn't on.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC889 has two SPDIF outputs: 0x06, 0x10. Board vendors can use either or both.
DX58SO uses 0x10, but the driver assumes 0x06. The safe solution is to add
0x10 as slave output to the existing 0x06.
Reported-by: Jeroen Van Breedam <jeroen.vanbreedam@sgr5.be>
Tested-by: Jeroen Van Breedam <jeroen.vanbreedam@sgr5.be>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added device id in struct for codec 92HD81B1C (0x111d76d5).
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a quirk model=acer-aspire for Acer Ferrari 5000 with ALC883 codec.
Note that model=auto doesn't work for this laptop because of broken BIOS
(that doesn't set the subsystem id properly).
Tested-by: Russ Dill <russ.dill@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace with the standard function calls to use caches for reading
the widget caps and pin caps.
hda_proc.c is still using the direct verbs to get raw values as
much as possible.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In patch_realtek.c, don't create empty or single-item "Input Source"
control elements that are simply superfluous.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The check for the amp-output must be done for widget-caps rather than
pin-caps as implemented in the recent change... Simply a thinko.
Also, add the similar checks to all places that put output-amp mutes
in the initialization.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added snd_hda_query_pin_caps() to read and cache pin-cap values
to avoid too frequently issuing the same verbs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't set amp-out values to pins without PINCAP_OUT capability,
which are usually assigned for digital mics on ALC663/ALC272.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch does two things:
Output Intel HDA Function Id in /proc/asound/cardX/codec#X
Align Vendor/Subsystem/Revision Ids to 8 characters, front-padded with zeros
Before:
Vendor Id: 0x11d41884
Subsystem Id: 0x103c281a
Revision Id: 0x100100
After:
Function Id: 0x1
Vendor Id: 0x11d41884
Subsystem Id: 0x103c281a
Revision Id: 0x0100100
As report on the Kernel Bugzilla #12888
Signed-off-by: Pascal de Bruijn <pascal@unilogicnetworks.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the detection of digital-mic inputs on ALC663 / ALC272 codecs
in the auto-detection mode. The automatic mic switch via plugging
isn't implemented yet, though.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, the prepare callback is called multiple times, BDL entries
are reset and re-programmed at each time.
This patch adds the check to avoid the reset of BDL entries when the
same parameters are used.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is an omitted unlock in one snd_mixart_hw_params fail path. Fix it.
Signed-off-by: Jiri Slaby <jirislaby@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If ratesp or formatsp values are zero, wrong values are passed to ALSA's
the PCM midlevel code. The bug is showed more later than expected.
Also, clean a bit the code.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The position-buffer on ATI controllers are unreliable as well as
on VIA chips, thus the same workaround for DMA position reading as
VIA is useful for ATI.
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ATI controllers (at least some SB0600 models) appear buggy to handle
64bit DMA. As a workaround, reset GCAP bit0 and let the driver to
use only 32bit DMA on these controllers.
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The previous commit breaks the (digital-) beep on ALC662.
ALC662 has the connection index 0x05 while ALC662 and ALC272 have the
index 0x04 for the beep widget.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC662/663 codecs have Beep Amplifier Index 0x04 not 0x05 in 0x0b NID.
Confirmed by testing on real hardware.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With this patch the drivers do not set the vmixer volume anymore at startup
because it is actually the output volume of the voices and ALSA mandates
that the volume must be 0 by default.
Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is a long standing bug in the drivers for cards with a vmixer because
I overlooked a detail in the c++ generic driver by echoaudio. Those cards
do not have a line-out volume control. It is a virtual control provided by
the generic driver. The bug is harmless because the DSP just ignores the
command to change the volume.
*NB:* It breaks alsa-tools/echomixer. A patch for it will follow.
This patch removes the line-out volume control from vmixer-equipped cards.
Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the power state of each widget before starting the initialization
work so that all verbs are executed properly.
Also, keep power-up during hwdep reconfiguration.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Update the places where the 0x1d widget is used for Conexant 5047, fixing
mismatch introduced after changing the connection.
Signed-off-by: Gregorio Guidi <gregorio.guidi@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clean up Conexant 5047 pareser code:
- Split mixer elements to separate arrays to reduce the duplicated
entires
- Fix mixer element names to the standard ones
- Remove unneeded cxt5047_hp2_unsol_event; the normal unsol_event
handler works fine.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the initial connections of output pins 0x13 and 0x1d for Conexant
5047 codec to point to the mixer amp properly.
Removed unneeded (doubly) verbs from arrays, also removed the unneeded
changing of widget 0x1c, which is now completely unused.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove superfluous verbs from cxt5047_toshiba_init_verbs[].
Also fix comments and minor coding style issues.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create "Capture Source" control dynamically for Conexant codecs.
If only one capture item is available, don't create such a control
since it's just useless.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of binding volumes, create a virtual master volume for Conexant
codecs. This allows separate HP and speaker volume controls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Code rework, comments of mail tiwai@suse.de (2009-03-09) incorporated.
Code tested on HP HDX16 (not tested on HDX18 yet).
Signed-off-by: Christoph Plattner <christoph.plattner@gmx.at>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added codec recognition of HP HDX platforms and added support of the
MUTE LED (orange/white). For this feature the CONFIG_SND_HDA_POWER_SAVE
is needed to use event handling for mute control.
Signed-off-by: Christoph Plattner <christoph.plattner@gmx.at>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the HT-Omega Claro (halo) sound cards, the headphone amplifier must
be enabled explicitly by setting a GPIO bit.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
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 - Fix headphone-detect regression with multiple HP jacks
ALSA: hda - Fix typos in slave controls in patch_sigmatel.c
Assign DACs to HP and speaker before mic-in/line-in shared outputs.
This improves the usability as it results in more intuitive mixer
names.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In stac92xx_auto_fill_dac_nids[], connect to the primary DAC if no
individual DAC is available for each pin. This ensures that the pin
works somehow at least.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create multiple "Headphone" and "Speaker" controls with non-zero index
numbers instead of "Headphone2", etc.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Improve the parser to pick up more intuitive control names for the
outputs judging from the pin type, instead of fixed names assigned
to channels.
Also, revive the multi-HP workaround since this change fixes the
problem with the multi-HP detection.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent changes over the DAC detection mechanism in patch_sigmatel.c
breaks the HP detection on the machines with multiple HP jacks.
It's basically because of the workaround to support the multi-channel
output. Since the HP detection is more important feature, disable
the HP-swap workaroud temporarily.
Reference: Novell bnc#482052
https://bugzilla.novell.com/show_bug.cgi?id=482052
Signed-off-by: Takashi Iwai <tiwai@suse.de>
"Headphone Playback ..." appears twice in slave_vols[] and slave_sws[].
They should be "Headphone Playback2 ..."
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Noises can be heard on analog outputs of (some model of) Lenovo
Ideapad due to the hardware problem, and the only workaround right now
is to fix the sample rate to 44.1kHz.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Ignore MIDI and PCM events in the interrupt handler until the device
gets initialized properly. Otherwise you may get kernel panic by the
access to uninitialized devices via hotplugging.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mute speaker outputs on headphone insertion for machines that use
3stack-hp model.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent update enabled the model=sony-assamd for all ALC262 with
PCI SSID 104d:90xx. But this includes the VAIO VGN-AR* that has the
primary codec of STAC92xx and the secondary ALC262 as a slave
digital-only codec. For this device, the model=auto must be chosen
to work properly.
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>
It's false positive, but annoying.
sound/pci/hda/hda_codec.c: In function ‘get_empty_pcm_device’:
sound/pci/hda/hda_codec.c:2772: warning: ‘dev’ may be used uninitialized in this function
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 probe_mask default for Toshiba laptop with ALC268
ALSA: hda - Add quirk for new HP xw series
ALSA: hda - Fix digital mic on dell-m4-1 and dell-m4-3
Allow more options to be set/reset via hwdep hint entry.
hp_detect, gpio_mask, gpio_dir, gpio_data, eapd_mask and eapd_switch
can be checked.
For example, to disable hp_detect on the fly,
# echo "hp_detect=0" > /sys/class/sound/hwC0D0/hints
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't create "Analog Loopback" controls as default since these controls
are usually more harmful than useful for normal users.
Only created when "loopback = yes" hint is given.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added snd_hda_get_hint() and snd_hda_get_bool_hint() helper functions
to retrieve a hint value.
Internally, the hint is stored in a pair of two strings, key and val.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't return a fatal error to the driver but continue to probe when
any error occurs at creating PCM streams for each codec.
It's often non-fatal and keeping it would help debugging.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Revert the codec probe instead of returning the error to the driver
when any error occurs at creating the control elements.
The control element conflict can be non-fatal in many cases,
especially if it comes from the digital-only codec.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Revert the Toshiba probe_mask quirk for 2.6.29 kernel
(commit 38f1df27e3).
In the current tree, the digital-only codec is handled properly so
no codec conflict should occur.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When an ALC268 codec is set up as the digital-only (as found in Toshiba
laptops), it shouldn't contain any beep control that conflict with the
primary codec.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some Toshiba laptops have another ALC268 codec on slot#3 that conflicts
with the primary codec. The codec#3 is for the digital I/O, and should
be fixed by the driver, but it'd need a bunch of changes.
So, let's fix the probe problem temporarily by setting the default
probe_mask value.
Reference: kernel bugzilla #12735http://bugzilla.kernel.org/show_bug.cgi?id=12735
Tested-by: Alexey Dobriyan <adobriyan@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Forgot to remove an unused variable.
sound/pci/hda/patch_realtek.c: In function ‘alc882_auto_init_analog_input’:
sound/pci/hda/patch_realtek.c:7018: warning: unused variable ‘vref’
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix num_dmuxes initialization for dell-m4-1 and dell-m4-3 models
of IDT 92HD71bxx codec, which was wrongly set to zero.
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Model drivers assume that model_data is zeroed, so we better use
kzalloc() (like we did before when it was allocated together with the
card structure).
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the Device IDs for MCP89 HD audio controller.
Removed the IDs of MCP7B cause this chipset had been cancelled.
Signed-off-by: Peer Chen <peerchen@gmail.com>
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>
Fix this sparse warning:
sound/pci/hda/hda_codec.c:1544:19: warning: incorrect type in assignment (different signedness)
sound/pci/hda/hda_codec.c:1544:19: expected unsigned long *vals
sound/pci/hda/hda_codec.c:1544:19: got long *<noident>
Signed-off-by: Hannes Eder <hannes@hanneseder.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix this sparse warnings:
sound/pci/emu10k1/emu10k1_main.c:723:66: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:724:68: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:748:74: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:751:66: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:759:73: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:760:73: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:837:50: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:845:50: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:881:50: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:889:57: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:890:57: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:895:60: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:897:60: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:899:60: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:910:56: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:914:57: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:918:56: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:922:57: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:924:58: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:936:60: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:1073:60: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:1088:60: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:1093:58: warning: incorrect type in argument 3 (different signedness)
Signed-off-by: Hannes Eder <hannes@hanneseder.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the pseudo device-locking using card->shutdown flag to avoid
the crash via clear/reconfig during operations.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set the upper limit 0dB to the volume of mixer amp 0x20 for
AD1984A HP laptops. The overloaded volume may damage the internal
speaker.
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>
sleeping for 2 seconds before checking for the iobox is not enough
on some systems.
this patch increases the timeout, but polls the card during that
time. it thus speeds up the module loading when the card has already
been initialized, while being more robust on systems, which require
a higher timeout than the predefined 2 seconds.
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Audiowerk2 driver snd-aw2 is bound to any saa7146 device as it does not
check subsystem ids. Many DVB devices are saa7146 based, so aw2 driver
grabs them as well.
According to http://lkml.org/lkml/2008/10/15/311 aw2 devices have the
subsystem ids set to 0, the saa7146 default.
Fix conflicts with DVB devices by checking for subsystem ids = 0
specifically.
Signed-off-by: Anssi Hannula <anssi.hannula@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the generic pincfg cache and save/restore functions.
Also introduced the pin-overriding via hwdep sysfs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use -60 dB as the minimum value of the master volume mixer control.
While the DACs would support ranges down to about -120 dB, such
attenuations are not useful in practice.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HT-Omega Claro halo's ADC is an AK5385 instead of a WM8785, so we
should handle the ADC parameters as we do with the X-Meridian.
Using the code for the wrong ADC does not seem to have any audible
effects, and the Windows driver does it, but it is nonetheless a good
idea to run the AK5385 with an oversampling ratio that is not outside
the documented limits.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The usage and comments make it clear values of 1/0 were intended
rather than -1/0
Noticed by sparse:
sound/pci/pcxhr/pcxhr.h💯20: error: dubious one-bit signed bitfield
sound/pci/pcxhr/pcxhr.h:101:22: error: dubious one-bit signed bitfield
sound/pci/pcxhr/pcxhr.h:102:24: error: dubious one-bit signed bitfield
sound/pci/pcxhr/pcxhr.h:103:21: error: dubious one-bit signed bitfield
sound/pci/pcxhr/pcxhr.h:104:25: error: dubious one-bit signed bitfield
sound/pci/pcxhr/pcxhr.h:105:20: error: dubious one-bit signed bitfield
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the EEPROM was partially overwritten (which seems to happen before the OS is
booted), restore its entire contents by deducing it from the remaining
information.
This does not have any effect on the Linux driver, which works even with
incomplete information in the EEPROM, but it makes other drivers work again.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Under as yet unknown circumstances, the first word of the sound card's
EEPROM gets overwritten. When this has happened, we cannot rely on the
subsystem IDs that the kernel reads from the PCI configuration
registers. Instead, we read the IDs directly from the EEPROM and do the
ID matching manually.
Because the model-specific driver cannot determine the model before
calling oxygen_pci_probe(), that function now gets a get_model()
callback as parameter. The customizing of the model structure, which
was formerly done by the probe() callback, also has moved into
get_model().
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When allocating resources, use a fixed name instead of reading it from
the model structure. This allows us to allocate the resources before
the actual model is known.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allocate the model-specific data dynamically instead of including it in
the memory block of the card structure. This will allow us to determine
the actual model after the card creation.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the owner field out of the oxygen_model structure and make it
a parameter of oxygen_pci_probe(), because the actual owner module does
not depend on the card model. Furthermore, moving it out of the model
structure allows us to create the card structure before the actual model
is known.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 7e86c0e685 ("do not
overwrite EEPROM on Xonar D2/D2X") because it did not actually help with
the problem.
More user reports show that the overwriting of the EEPROM is not
triggered by using this driver but by installing Linux, and that the
installation of any other operating system (even one without any CMI8788
driver) has the same effect. In other words, the presence of this
driver does not have any effect on the occurrence of the error. (So
far, the available evidence seems to point to a BIOS bug.)
Furthermore, it turns out that the EEPROM chip is protected against
stray write commands by the command format and by requiring a separate
write-enable command, so the error scenario in the previous commit (that
SPI writes can be misinterpreted as an EEPROM write command) is not even
theoretically possible.
The mixer control that was removed as a consequence of the previous
commit can only be partially emulated in userspace, which also means it
cannot be seen be the in-kernel OSS API emulation, so it is better to
revert that change.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC268 can be configured as digital-only, e.g. for HDMI, on some
machines. Allow the parser to set up the digital-only mode.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed a typo of || and &&.
As it's in a disabled code section, there is no behavior change, though.
Reported-by: Jörg-Volker Peetz <jvpeetz@web.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Force speaker pin config with model=hp-dv5 model for cases when bios
doesn't set it up properly. All reported hp laptops using model=hp-dv5
model have speaker at pin 0x0d with same config, so it's safe to add
this within hp-dv5 model.
Reference: alsa-devel mailing list thread on
http://mailman.alsa-project.org/pipermail/alsa-devel/2009-February/014390.html
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 32e176c14d.
That commit caused a regression with suspend on Thinkpad SL300.
Reference: kernel bug#12711
http://bugzilla.kernel.org/show_bug.cgi?id=12711
Tested-by: Alexandre Rostovtsev <tetromino@gmail.com>
Acked-by: Rafael J. Wysocki <rjw@sisk.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the support of multiple digital outputs via auto-probing for
Realtek ALC88x codecs. The multiple outputs are handled as slave
streams, so only one PCM stream (and the corresponding IEC958*
elements) is provided to control both digital outputs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the helper function snd_hda_multi_out_dig_cleanup() to clean up
the digital outputs with multi setup. This call is needed in cases
the codec supports multiple digital outputs as slaves. Otherwise the
slave widgets aren't properly cleaned up.
For a single digital output (e.g. in patch_conexant.c), this call isn't
needed.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Force model=auto for Acer AX1700-U3700A with ALC888 codec.
Since Acer devices are handlded as model=acer as default, the auto
parsing has to be specified explicitly.
(Maybe it's better rather to remove this default model=acer handling,
though.)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change HP dv7 quirk: although reported to work with hp-m4 model
(https://bugzilla.novell.com/show_bug.cgi?id=445321), the original
report doesn't contain info about testing of internal microphone.
Recently I received a report about internal mic not working
(https://qa.mandriva.com/show_bug.cgi?id=44855#c193), this must be
related with the forced line in on pin 0x0e done with hp-m4 model. Thus
change the current quirk from STAC_HP_M4 to STAC_HP_DV5, later reported
to be fixed on a provided kernel with this change
(https://qa.mandriva.com/show_bug.cgi?id=44855#c196).
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
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>
Add STAC_DELL_S14 quirk for new laptop series. Removed un-needed pins
in pin_nids for stac92hd83xxx. Also reorganized connection selection
code for the respective ports per quirk define.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASUS W5F needs the fixed codec-slots to probe to override the BIOS
problem.
Tested-by: Giovanni Moser Frainer <giovanni@redix.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some devices have broken BIOS and they don't set the codec probe-bit
properly after cleared by the driver. This makes the driver skipping
the necessary codec slots.
Since BIOS update isn't always easy, now the semantics of probe_mask
option is changed a bit. When it contains the bit 8 (0x100), the
lower bits are used to probe that slots regardless of codec-probe bits
returned by the hardware.
For example, probe_mask=0x103 will force to probe the codec slot #0
and #1.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After commit "ALSA: hda - Fix restore of pin configs at resume for
STAC/IDT codecs", the introduced stac_save_pin_cfgs function checks
already for pins == NULL case, saving then default pin configs from
machine with stac92xx_save_bios_config_regs. So we can remove the
extra checks when stac927x_brd_tbl[spec->board_config] == NULL.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The devices that have been newly added during reconfig must be
registered. Otherwise they won't be visible to user-space.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Detect multiple digital-out pins in snd_hda_parse_pin_defconfig().
The dig_out_pin and dig_out_type fields become arrays.
The codec parser still doesn't use this multiple pins detection, though.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We found that enabling/disabling HDMI audio pin out at stream start/stop
time will kill the leading 500ms or so sound samples. Avoid this by enabling
pin out once and for ever at module loading time.
The leading ~500ms audio samples will still be lost when switching from
X-channel playback to Y-channel playback where X != Y. However there's no
much we can do about it: the audio infoframe has to change and it looks like
either G45 or YAMAHA requires some time to switch the configuration.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The YAMAHA AV-X1800 requires audio infoframe to include speaker-channel
mapping to play >2 channel HDMI audio. In theory that mapping should be
derived from its speaker configurations contained in its ELD. However we
currently cannot get ELD in console before the KMS functionalities are ready.
This is a more or less general issue at least in the near future. As a
workaround, we propose to allow playback of mult-channel audio when ELD
is not available.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the digital beep support for ALC268. It was missing in the
last patches.
However, ALC268 has a strange pin use for widget 0x1d, which could be
used as another purpose. So, the patch adds a check of the beep control
before creating the hook for input layer.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When checking for input amps on pins 0x0a, 0x0d and 0x0f, and
initializing them for 92hd71xxx codec models, we must skip nid 0x0f
for 4-port models too like with 5-port models, as it is unused
(nid 0x0f is vendor reserved in 4-port models).
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds a new sound quirk entry (model=ecs202) for an ecs motherboard
with IDT STAC9221 codec (1019:2950).
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use digital beep instead of analog pc-beep for AD codecs.
Create the beep mixer controls dynamically on demand.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the SPDIF pin as slave digital out to enable concurrent
HDMI/SPDIF outputs for ASUS M3A-H/HDMI with ALC1200 codec.
Tested-by: Thomas Schneider <nailstudio@gmx.net>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the ADATOut nid to a slave digital outs struct to allow output
via the DigOut pin.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't use uneeded/wrong third parameter for stac92xx_parse_auto_config
in patch_stac92hd71bxx (no SPDIF in).
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Detect the number of connected ports and number of smuxes dynamically,
looking at pin configs, using new introduced functions
stac92hd71bxx_connected_ports and stac92hd71bxx_connected_smuxes. Also
use proper input mux configuration for 4port and 5port models.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current code for STAC92HD71Bx and STAC92HD75Bx doesn't consider pin
complexes 0x20 and 0x27. Also for 4 port models, nids 0x0e and 0x0f
are vendor reserved. This commit changes code so it'll consider the
additional pin complexes for models that have it, and avoid reserved
nids to be touched on 4 port models.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some fixes regarding snd-hda-intel workqueue:
- Use create_singlethread_workqueue() instead of create_workqueue()
as per-CPU work isn't required.
- Allocate workq name string properly
- Renamed the workq name to "hd-audio*" to be more obvious.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A digital beep generator can be used via input layer.
Signed-off-by: Kusanagi Kouichi <slash@ma.neweb.ne.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Realtek ALC262 on the Tyan Thunder n6650W (S2915-E) mainboard has a
rather odd configuration template. As a result, the white AUX connector
can not be used. This rewrites the default config to more accurately
reflect the connector layout, colour and function.
Unfortunately the black CD_IN connector, which is suspected to be widget
0x1c refuses to switch into input (0x20), instead opting to remain on 0x0.
As such, no mixer controls are exposed for it. Autoswitching is implemented
between the front headphone output and back line output.
Signed-off-by: Tony Vroon <tony@linx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
MIDI_TX IRQ seems always pending when any bytes on FIFO is available.
Thus, it's better to enable MPU_TX only when any bytres are really
stored in the substream, and disables immediately when the queue
becomes empty.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Volume-knob widgets have no widget selection although they have widget
connections. Thus, the connection list in the proc output shouldn't
contain the selection (*).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With a postfix decrement time will reach -1 rather than 0,
so the warning will not be issued.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the case of having a selector instead of mixer while initing input
sources, the case that happens with ALC889, we must select instead
of muting input. Problem was found while testing with hda-emu.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While debugging some code paths in AC97 codec patches and its suspend
and resume functions, getting to know the flags has proved useful to
follow those code paths.
Signed-off-by: Thadeu Lima de Souza Cascardo <cascardo@holoscopio.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently code for capture source support in ALC889 only considers
capture mixers. This change adds additional support for ADC+selector
present in ALC889, taking into account also the presence of an
additional DMIC connection item in the selector.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check in a quirk list if it should do cold reset when AC97 power saving
is enabled. Some devices do not resume properly when cold reset,
although power saving works OK.
Signed-off-by: Thadeu Lima de Souza Cascardo <cascardo@holoscopio.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add another MacBook Pro 4,1 SSID (106b:3800). It seems that latter revisions,
(at least mine), have different IDs to earlier revisions.
Signed-off-by: Luke Yelavich <themuso@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some revisions of the 92hd71xxx codec families don't have input amps
on ports 0xa, 0xd and 0xf, so probe the widget caps on port 0xa and
check for support, if found run snd_hda_sequence_write_cache() on the
stac92hd71xxx_unmute_core_init verb list.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- add support for gpio ports (2 GPI, 2 GPO) of pcxhr stereo cards
- minor bugfixes : allow setting clock to internal by the mixer
even if there is no stream (but monitoring)
Signed-off-by: Markus Bollinger <bollinger@digigram.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The mic-boosts (0x14 and 0x15) on AD1884* codecs are input-amps,
not output-amps. Fix the invalid initialization verbs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of fixed kcontrol_new element, build "Input Source" controls
dynamically. If the number of input-source items is 0 or 1, we don't
need to create such a control.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The reference NID for the analog outputs of STAC/IDT codecs is set
to a fixed number 0x02. But this isn't always correct and in many
codecs it points to a non-existing NID.
This patch fixes the initialization of the PCM reference NID taken
from the actually probed DAC list.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
