BugLink: https://launchpad.net/bugs/683695
The original reporter states that headphone jacks do not appear to
work. Upon inspecting his codec dump, and upon further testing, it is
confirmed that the "alienware" model quirk is correct.
Reported-and-tested-by: Cody Thierauf
Cc: <stable@kernel.org> [2.6.32+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for the RME HDSP RPM IO box. Changes have been made in the identification of the IO box and the neccessary controls have been added.
Signed-off-by: Florian Faber <faberman@linuxproaudio.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/682199
A 2.6.35 (Ubuntu Maverick) user, burningphantom1, reported a regression
in audio: playback was inaudible through both speakers and headphones.
In commit 272a527c04 of sound-2.6.git, a new model was added with this
machine's PCI SSID. Fortunately, it is now sufficient to use the auto
model for BIOS auto-parsing instead of the existing quirk.
Playback, capture, and jack sense were verified working for both
2.6.35 and the alsa-driver snapshot from 2010-11-27 when model=auto is
used.
Reported-and-tested-by: burningphantom1
Cc: <stable@kernel.org> [2.6.35+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When SKU assid gives no valid bits for 0x38, the driver didn't take
any action, so far. This resulted in the missing initialization for
external amps, etc, thus the silent output in the end.
Especially users hit this problem on ALC888 newly since 2.6.35,
where the driver doesn't force to use ALC_INIT_DEFAULT any more.
This patch sets the default initialization scheme to use
ALC_INIT_DEFAULT when no valid bits are set for SKU assid.
Reference:
https://bugzilla.redhat.com/show_bug.cgi?id=657388
Reported-and-tested-by: Kyle McMartin <kyle@redhat.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The mixer nids passed to alc_auto_create_input_ctls are wrong: 0x15 is
a pin, and 0x09 is the ADC on both ALC660-VD/ALC861-VD. Thus with
current code, input playback volume/switches and input source mixer
controls are not created, and recording doesn't work. Select correct
mixers, 0x0b (input playback mixer) and 0x22 (capture source mixer).
Reference: https://qa.mandriva.com/show_bug.cgi?id=61159
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch enables ALC887-VD to use the DAC at nid 0x26,
which makes it possible to use this DAC for e g Headphone
volume.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The refactoring commit d433a67831
ALSA: hda - Optimize the check of ALC269 codec variants
introduced a wrong check for ALC269-vb type. This patch corrects it.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a quirk entry for Thinkpad Edge 11 as well as other TP Edge models.
Signed-off-by: Manoj Iyer <manoj.iyer@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a generic callback function for fixup elements. This can be used
to do some unusual things like overriding the AMP cache, etc.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC887-VD is like ALC888-VD. It can not be initialized as ALC882.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/677652
The original reporter states that, in 2.6.35, headphones do not appear
to work, nor does inserting them mute the A52J's onboard speakers. Upon
inspecting the codec dump, it appears that the newly committed hp-laptop
quirk will suffice to enable this basic functionality. Testing was done
with an alsa-driver build from 2010-11-21.
Reported-and-tested-by: Joan Creus
Cc: <stable@kernel.org> [2.6.35+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
. Fix PulseAudio "ALSA driver bug" issue
(if we have two alternated areas within a 64k DMA buffer, then max
period size should obviously be 32k only).
Back references:
http://pulseaudio.org/wiki/AlsaIssueshttp://fedoraproject.org/wiki/Features/GlitchFreeAudio
. In stop timer function, need to supply ACK in the timer control byte.
. Minor log output correction
When I did my first PA testing recently, the period size bug resulted
in quite precisely observeable half-period-based playback distortion.
PA-based operation is quite a bit more underrun-prone (despite its
zero-copy optimizations etc.) than raw ALSA with this rather spartan
sound hardware implementation on my puny Athlon.
Note that even with this patch, azt3328 still doesn't work for both
cases yet, PA tsched=0 and tsched
(on tsched=0 it will playback tiny fragments of periods, leading to tiny
stuttering sounds with some pauses in between, whereas with
timer-scheduled operation playback works fine - minus some quite increased
underrun trouble on PA vs. ALSA, that is).
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/677830
The original reporter states that the subwoofer does not mute when
inserting headphones. We need an entry for his machine's SSID in the
subwoofer pin fixup list, so add it there (verified using hda_analyzer).
Reported-and-tested-by: i-NoD
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow disabling period wakeup interrupts for all PCM streams.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/677652
The original reporter states that, in 2.6.35, headphones do not appear
to work, nor does inserting them mute the A52J's onboard speakers. Upon
inspecting the codec dump, it appears that the newly committed hp-laptop
quirk will suffice to enable this basic functionality. Testing was done
with an alsa-driver build from 2010-11-21.
Reported-and-tested-by: Joan Creus
Cc: <stable@kernel.org> [2.6.35+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
. Fix PulseAudio "ALSA driver bug" issue
(if we have two alternated areas within a 64k DMA buffer, then max
period size should obviously be 32k only).
Back references:
http://pulseaudio.org/wiki/AlsaIssueshttp://fedoraproject.org/wiki/Features/GlitchFreeAudio
. In stop timer function, need to supply ACK in the timer control byte.
. Minor log output correction
When I did my first PA testing recently, the period size bug resulted
in quite precisely observeable half-period-based playback distortion.
PA-based operation is quite a bit more underrun-prone (despite its
zero-copy optimizations etc.) than raw ALSA with this rather spartan
sound hardware implementation on my puny Athlon.
Note that even with this patch, azt3328 still doesn't work for both
cases yet, PA tsched=0 and tsched
(on tsched=0 it will playback tiny fragments of periods, leading to tiny
stuttering sounds with some pauses in between, whereas with
timer-scheduled operation playback works fine - minus some quite increased
underrun trouble on PA vs. ALSA, that is).
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/677830
The original reporter states that the subwoofer does not mute when
inserting headphones. We need an entry for his machine's SSID in the
subwoofer pin fixup list, so add it there (verified using hda_analyzer).
Reported-and-tested-by: i-NoD
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://launchpad.net/bugs/669092
ALC887 does not have any volume control ability on the mixer NIDs,
so put the volume controls on the dac NIDs instead. Without this
patch, ALC887 users cannot use alsamixer at all.
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/669279
The original reporter states: "The Master mixer does not change the
volume from the headphone output (which is affected by the headphone
mixer). Instead it only seems to control the on-board speaker volume.
This confuses PulseAudio greatly as the Master channel is merged into
the volume mix."
Fix this symptom by applying the hp_only quirk for the reporter's SSID.
The fix is applicable to all stable kernels.
Reported-and-tested-by: Ben Gamari <bgamari@gmail.com>
Cc: <stable@kernel.org> [2.6.32+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In each function, the value apcm is stored in the private_data field of
runtime. At the same time the function ct_atc_pcm_free_substream is stored
in the private_free field of the same structure. ct_atc_pcm_free_substream
dereferences and ultimately frees the value in the private_data field. But
each function can exit in an error case with apcm having been freed, in
which case a subsequent call to the private_free function would perform a
dereference after free. On the other hand, if the private_free field is
not initialized, it is NULL, and not invoked (see snd_pcm_detach_substream
in sound/core/pcm.c). To avoid the introduction of a dangling pointer, the
initializations of the private_data and private_free fields are moved to
the end of the function, past any possible free of apcm. This is safe
because the previous calls to snd_pcm_hw_constraint_integer and
snd_pcm_hw_constraint_minmax, which take runtime as an argument, do not
refer to either of these fields.
In each function, there is one error case where apcm needs to be freed, and
a call to kfree is added.
The sematic match that finds this problem is as follows:
(http://coccinelle.lip6.fr/)
// <smpl>
@@
expression e,e1,e2,e3;
identifier f,free1,free2;
expression a;
@@
*e->f = a
... when != e->f = e1
when any
if (...) {
... when != free1(...,e,...)
when != e->f = e2
* kfree(a)
... when != free2(...,e,...)
when != e->f = e3
}
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the platform already provides a definition for these accessors
do not redefine them. The warning was caught on MIPS.
Signed-off-by: Florian Fainelli <florian@openwrt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://launchpad.net/bugs/673075
According to the datasheet of 92HD87B, there is a digital mic
at nid 0x11, so enable it in order to be able to use the mic.
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
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>
Sort the PCI IDs so that they make logical sense. Also move the card
name comments into this list because the model symbols should be (more)
self-explanationary.
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>
This patch add support for the MacBookAir3,1 and MacBookAir3,2 to the alsa
sound system.
Signed-off-by: Edgar (gimli) Hucek <gimli@dark-green.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I noticed that sound/pci/asihpi/hpicmn.c::hpi_alloc_control_cache() does
not check the return value from kmalloc(), which may fail.
If kmalloc() fails we'll dereference a null pointer and things will go bad
fast.
There are two memory allocations in that function and there's also the
problem that the first may succeed and the second may fail and nothing is
done about that either which will also go wrong down the line.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Acked-by: Eliot Blennerhassett <linux@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
converts a 1 bit signed bitfield to an unsigned.
Reported-by: Dr. David Alan Gilbert <linux@treblig.org>
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When reading through sound/pci/cs46xx/dsp_spos.c I noticed a couple of
things in cs46xx_dsp_spos_create().
It seems to me that we don't always free the various memory buffers we
allocate and we also do some work (structure member assignment) early,
that is completely pointless if some of the memory allocations fail and
we end up just aborting the whole thing.
I don't have hardware to test, so the patch below is compile tested only,
but it makes the following changes:
- Make sure we always free all allocated memory on failures.
- Don't do pointless work assigning to structure members before we know
all memory allocations, that may abort progress, have completed
successfully.
- Remove some trailing whitespace.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Tested-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some HP laptops have lower amplifier levels for speakers in comparison
with headphone outputs. This patch changes the BTL amp level for these
machines to balance both the speaker and headphone output levels.
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.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: (365 commits)
ALSA: hda - Disable sticky PCM stream assignment for AD codecs
ALSA: usb - Creative USB X-Fi volume knob support
ALSA: ca0106: Use card specific dac id for mute controls.
ALSA: ca0106: Allow different sound cards to use different SPI channel mappings.
ALSA: ca0106: Create a nice spot for mapping channels to dacs.
ALSA: ca0106: Move enabling of front dac out of hardcoded setup sequence.
ALSA: ca0106: Pull out dac powering routine into separate function.
ALSA: ca0106 - add Sound Blaster 5.1vx info.
ASoC: tlv320dac33: Use usleep_range for delays
ALSA: usb-audio: add Novation Launchpad support
ALSA: hda - Add workarounds for CT-IBG controllers
ALSA: hda - Fix wrong TLV mute bit for STAC/IDT codecs
ASoC: tpa6130a2: Error handling for broken chip
ASoC: max98088: Staticise m98088_eq_band
ASoC: soc-core: Fix codec->name memory leak
ALSA: hda - Apply ideapad quirk to Acer laptops with Cxt5066
ALSA: hda - Add some workarounds for Creative IBG
ALSA: hda - Fix wrong SPDIF NID assignment for CA0110
ALSA: hda - Fix codec rename rules for ALC662-compatible codecs
ALSA: hda - Add alc_init_jacks() call to other codecs
...
The sticky PCM stream assignment introduced in 2.6.36 kernel seems
causing problems on AD codecs. At some time later, the streaming no
longer works by unknown reason. A simple workaround is to disable
sticky-assignment for these codecs.
Tested-by: Vasily Khoruzhick <anarsoul@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is to allow a future patch to have card specific mappings between
dacs, which is required since the Sound Blaster 5.1vx seems to have a
different mapping to what was previously used.
Signed-off-by: Andy Owen <andy-alsa@ultra-premium.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is ground work for a future commit where cards (such as the Sound
Blaster 5.1vx) have different mappings between dacs and channels.
Signed-off-by: Andy Owen <andy-alsa@ultra-premium.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Creative IBG controllers require the playback stream-tags to be started
from 1, instead of capture+1. Otherwise the stream stalls.
Reported-by: Wai Yew CHAY <wychay@ctl.creative.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The bit value set for TLV mute was wrong in commit
de8c85f784, which resulted in bogus
dB ranges that screw up PulseAudio. Corrected with the right constant.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Multiple Acer laptops with the SSID 1025:04xx require the quirk
mode=ideapad, so let's use mask to apply to all these.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Creative HD-audio controller chips require some workarounds:
- Additional delay before RIRB response
- Set the initial RIRB counter to 0xc0
The latter seems to be done in general in Windows driver, so we may
use this value later for all types if it's confirmed to work better.
Reported-by: Wai Yew CHAY <wychay@ctl.creative.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The dig_out_nid field must take a digital-converter widget, but the current
ca0110 parser passed the pin wrongly instead.
Reported-by: Wai Yew CHAY <wychay@ctl.creative.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Windows may leave pin power-down registers set after reboot, and
this resulted in muted output on Linux. Reset these registers
at initialization properly.
Signed-off-by: Charles Chin <Charles.Chin@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch below updates broken web addresses in the kernel
Signed-off-by: Justin P. Mattock <justinmattock@gmail.com>
Cc: Maciej W. Rozycki <macro@linux-mips.org>
Cc: Geert Uytterhoeven <geert@linux-m68k.org>
Cc: Finn Thain <fthain@telegraphics.com.au>
Cc: Randy Dunlap <rdunlap@xenotime.net>
Cc: Matt Turner <mattst88@gmail.com>
Cc: Dimitry Torokhov <dmitry.torokhov@gmail.com>
Cc: Mike Frysinger <vapier.adi@gmail.com>
Acked-by: Ben Pfaff <blp@cs.stanford.edu>
Acked-by: Hans J. Koch <hjk@linutronix.de>
Reviewed-by: Finn Thain <fthain@telegraphics.com.au>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
The HDA specification does not allow for a codec to mute itself just
because the volume is reduced, so _of course_ somebody had to go and do
it. This wouldn'\''t hurt too much when the volume is adjusted by hand,
but programs like PA that try to set the volume automatically could
inadvertently mute the output.
To work around this, change the TLV dB information for the Master volume
on all Sigmatel HDA codecs to indicate the the minimal volume setting
actually mutes.
Reported-by: Colin Guthrie <gmane@colin.guthr.ie>
Reported-by: "Alexander E. Patrakov" <patrakov@gmail.com>
Tested-by: Colin Guthrie <cguthrie@mandriva.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://launchpad.net/bugs/617647
The current SKU value disables playback, so ignore the SKU value.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Realtek have ways of specifying external amps and more via a
special nid or via the Codec's subsystem ID, this is called "SKU".
The computer manufacturer sometimes gets this wrong, so we need
to be able to override or ignore the SKU customization value.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.launchpad.net/bugs/653420
Add another HP DV6 notebook (103c:363e) to use STAC_HP_DV5.
Signed-off-by: Luke Yelavich <luke.yelavich@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This board has a strange PCI SSID 13f6:ffff. Works as compabile as
MODEL_CMEDIA_REF.
Reported-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Configure the PEX8111 bridge on the PCI Express cards so that the audio
DMA controller can do proper burst reads and is less likely to lose
data. This is usually done automatically, but is required on older
cards where the user has not applied the PLX firmware update.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the PCIe/PCI bridge initialization code to configure only the
bridge that is actually connected to the sound chip, instead of any
bridge found in the system. The new code also makes it easier to add
other bridges.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The interrupt counter is independent of the buffer counter, so there are
no restrictions on the period size. Having fewer periods also makes
PulseAudio happy.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The input monitor half volume bit results in a factor of 0.5, so the
minimum scale value should be -6 dB.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rename the symbol for the XCID pins, fix up a decimal/hex confusion for
the CMI8787 package ID, and add the other known package IDs.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The anti-pop delay for the STX should be 800 ms, not 100 ms like the ST.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a PCI ID for the Xonar HDAV1.3 Slim. There is no actual support,
but the presence of the ID allows the EEPROM repair code to work for
this card.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are more models without a CD input than with one, so handle this
explicitly with a device_config flag to avoid having to define a control
filter callback to filter it out.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The controller on the Xonar DS is labeled "AV66", not "AV200".
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>
The SNDRV_HDSP_IOCTL_GET_CONFIG_INFO and
SNDRV_HDSP_IOCTL_GET_CONFIG_INFO ioctls in hdspm.c and hdsp.c allow
unprivileged users to read uninitialized kernel stack memory, because
several fields of the hdsp{m}_config_info structs declared on the stack
are not altered or zeroed before being copied back to the user. This
patch takes care of it.
Signed-off-by: Dan Rosenberg <dan.j.rosenberg@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix bug in switching between dmic and mic when both use the same mux.
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The SPDIF in audio widget must be searched through the list as the widget
that contains the given pin as the connection source. The current code
was implemented in a reverse way.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I've found the following patch is necessary to enable line-in on
my MacBookPro 5,3 machine. With the patch applied I've successfully
recorded audio from the line-in jack. This is based on the existing
5,5 support.
Signed-off-by: Vince Weaver <vweaver1@eecs.utk.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the HT-Omega Claro halo card, the ADC data must be captured from the
second I2S input. Using the default first input, which isn't connected
to anything, would result in silence.
Signed-off-by: Erik J. Staab <ejs@insightbb.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've applied a fix-up for ALC269 VAIO only for two models. But all
Sony VAIO models with ALC269 codec seem to require the similar fix.
Let's apply it with vendor-id mask.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The headphone and external-mic pin NIDs can be null, and the jack input
elements should be skipped in such a case.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace the explicit ifdef check and call of check_power_status ops with
a new helper function, hda_call_check_power_status().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Channel 2 and channel 3 were all wrongly mapped to HDMI slot 4.
This shows up as a bug that one channel is "lost" when playing in
surround41 mode.
Signed-off-by: Jerry Zhou <jerry.zhou@intel.com>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
DisplayPort works mostly in the same way as HDMI, except that it expects
a slightly different audio infoframe format.
Citations from "HDA036-A: Display Port Support and HDMI Miscellaneous
Corrections":
The HDMI specification defines a data island packet with a header of 4
bytes (3 bytes content + 1 byte ECC) and packet body of 32 bytes (28
bytes content and 4 bytes ECC). Display Port specification on the other
hand defines a data island packet (secondary data packet) with header of
4 bytes protected by 4 bytes of parity, and data of theoretically up to
1024 bytes with each 16 bytes chunk of data protected by 4 bytes of
parity. Note that the ECC or parity bytes are not present in the DIP
content populated by software and are hardware generated.
It tests DP connection based on the ELD conn_type field, which will be
set by the graphics driver and can be overriden manually by users
through the /proc/asound/card0/eld* interface.
The DP infoframe is tested OK on Intel SandyBridge/CougarPoint platform.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create a helper function to simplify the code.
Also, cleaned up the ifdef SND_HDA_NEEDS_RESUME and
CONFIG_SND_HDA_POWER_SAVE. The former is always defined when the latter
is set.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC269vb and other variants don't use the widgets 0x24 but prefer the
widget 0x22 instead. We need to fix the input parser.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't call the COEF check for checking ACL269 codec variants at each
time in init but remember the type at the initialization.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When quirks are applied, the numbers of output pins in autocfg aren't
set up properly but only pin arrays are changed. Let's fix it up so that
the rest of the parser can use autocfg.line_outs & co safely.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch merges all three patch_*hdmi variants to the single HDMI
parser. There is only one snd-hda-codec-hdmi module now.
In this patch, the behavior of each parser isn't changed much.
The old ATI parser still doesn't use the dynamic parser yet.
In later patches, they'll be cleaned up.
Also, this patch gets rid of the individual snd-hda-eld module and
builds into snd-hda-codec-hdmi, since this is referred only from the
HDMI parser.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The external mic jack for auto-mic switch must be really an external
jack and with a presense-detection capability. This patch makes the
check more paranoia.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make the helper function to give the input-pin attribute for jack
connectivity and location. This simplifies checks of input-pin jacks
a bit in some places.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Through the transition of autocfg to individual inputs array, I forgot
to rewrite the argument passed to alc_set_input_pin(). This resulted in
wrongly setup input pins. Fixed now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://launchpad.net/bugs/640254
In some cases a magic processing coefficient is needed to enable
the internal speaker on Dell M101z. According to Realtek, this
processing coefficient is only present on ALC269vb.
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Most of Intel controllers work as generic HD-audio without quirks,
and it'll be hopefully so in future. Let's mark pci id with the
PCI_CLASS_MULTIMEDIA_HD_AUDIO for Intel so that the driver will work
with any new control chips in future.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the preliminary support for new Conexant audio codecs with
14f1:5097, 14f1:5098, 14f1:50a1, 14f1:50a2, 14f1:50ab, 14f1:50ac,
14f1:50b8 and 14f1:50b9.
Unlike other Conexant parsers, this is designed to be mostly automatic,
parsing from BIOS pin configurations.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For avoiding the click noises at power-saving, set some COEF values
for ALC269* codecs.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a quirk for laptop Toshiba Satellite C650D to have proper external HP and
external Mic support.
Signed-off-by: Anisse Astier <anisse@astier.eu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the mic pins are assigned to the same location, we can omit the
redundant location prefix like "Front" or "Rear".
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch improves the input-source label strings to be generated from
the pin information instead of fixed strings per AUTO_PIN_* type.
This gives more suitable labels, especially for mic and line-in pins.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We can assign multiple pins to a single role now, let's reduce the
redundant FRONT_MIC and FRONT_LINE. Also, autocfg->input_pins[] is
no longer used, so this is removed as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Keep char array in the input_mux item itself instead of pointing to
an external string. This is a preliminary work for improving the
input-mux name based on the pin role.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Update the Xonar config texts with the latest information about the
Xonar DS, HDAV1.3 Slim, and Xense.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As the select directive does not handle indirect dependencies, select
those explicitly in the driver sections.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the possibility to route a mix of the two channels of stereo data to
the center and LFE outputs. Due to a WM8766 restriction, all surround
and back channels also get the mixed L/R signal in this case.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Automatically mute the speaker outputs as long as a headphone is plugged.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that the polarity of the headphone detection pin is known, replace
the debugging message with a proper jack plug input device.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the correct number, register bits, and names for the input switches.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The check for the volume update latch bit was accidentally in the wrong
function, where it would prevent the MSB from being written, instead of
correctly ignoring it for cached values.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
By adding the subwoofer as a speaker pin, it is treated correctly when auto-muting.
BugLink: https://launchpad.net/bugs/611803
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a fixup table for ALC262 codec containing the entry for FSC
Celsius H270. Now both headphone jacks are detected properly as
headphones.
Reference: Novell bnc637263
https://bugzilla.novell.com/show_bug.cgi?id=637263
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch changes the alc262 auto-parser to allow multiple pins
assigned for a single purpose (line-out, headphone or speaker).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently headphone auto-mute using alc_automute_pin() assumes only
the single pin used for the headphone output. Since there are devices
with multiple headphone jacks, we need to check all these pins there,
too.
Also this patch merges the common code between alc_automute_pin() and
alc_automute_amp() helper functions.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In snd_hda_parse_def_config(), some unused values may remain in hp_pins[]
array during the headphone-reassignment workaround. This patch clears
the unused array members.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_hda_parse_pin_def_config() has some workaround for re-assigning
some pins declared as headphones to line-outs. This didn't work properly
for some cases because it used memmove() stupidly wrongly.
Reference: Novell bnc#637263
https://bugzilla.novell.com/show_bug.cgi?id=637263
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Changing the way the input controls are named using port connection
type and jack location info.
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Daniel T Chen
Florian Faber
Takashi Iwai
Herton Ronaldo Krzesinski
David Henningsson
Denis Kuplyakov
Kailang Yang
Manoj Iyer
Andreas Mohr
Clemens Ladisch
Joe Perches
Julia Lawall
Florian Fainelli
Edgar (gimli) Hucek
Jesper Juhl
Uwe Kleine-König
Tim Blechmann
Linus Torvalds
Vitaliy Kulikov
Andy Owen
Charles
Justin P. Mattock
Andrea Gelmini
Luke Yelavich
Valentine Sinitsyn
Otavio Salvador
Dan Rosenberg
Vince Weaver
Erik J. Staab
Jerry Zhou
Wu Fengguang
Charles Chin
Stephen Warren
Anisse Astier
Seth Heasley