Merge fixes for Thinkpad docking-station regressions for 3.3 kernels
back to 3.4. These were committed in that branch to make the stable
merging easier.
Conflicts:
sound/pci/hda/patch_conexant.c
ThinkPad 410,420,510,520 and X201 with cx50585 & co chips have the
docking-station ports, but BIOS doesn't initialize for these pins.
Thus, like the former X200, we need to set up the pins manually in the
driver.
The odd part is that the same PCI SSID is used for X200 and T400, thus
we need to prepare individual fixup tables for cx5051 and others.
Bugzilla entries:
https://bugzilla.redhat.com/show_bug.cgi?id=808559https://bugzilla.redhat.com/show_bug.cgi?id=806217https://bugzilla.redhat.com/show_bug.cgi?id=810697
Reported-by: Josh Boyer <jwboyer@redhat.com>
Reported-by: Jens Taprogge <jens.taprogge@taprogge.org>
Tested-by: Jens Taprogge <jens.taprogge@taprogge.org>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some output pins on Conexant chips have no HP control bit, but the
auto-parser initializes these pins unconditionally with PIN_HP.
Check the pin-capability and avoid the HP bit if not supported.
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While refactoring the mute-LED handling for HP laptops, I messed up
the polarity check in a wrong way. The red (or the mute-LED if any)
should appear in the muted state, corresponding to GPIO on.
Reported-by: Mikko Vinni <mmvinni@yahoo.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Through the transition to the auto-parser, the support for
Quanta/Gericom KN1 got broken. There are two problems behind it:
- This machine doesn't like the default COEF setup for ALC260 we take
now as default
- BIOS doesn't set the pins correctly at all; especially the machine
uses only the pin 0x0f for both headphone and speaker
This patch adds the fixup as a workaround for these issues.
Reported-and-tested-by: Uros Vampl <mobile.leecher@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In sound/pci/hda/patch_realtek.c::alc_auto_fill_dac_nids(), in the
'for (;;)' loop, if the 'badness' value returned from
fill_and_eval_dacs() is negative, then we'll return from the function
without freeing the memory we allocated for 'best_cfg', thus leaking.
Fix the leak by kfree()'ing the memory when badness is negative.
While I was there I also noticed some trailing whitespace in the
function that I removed (along with all other trailing whitespace in
the file) - it didn't seem worth-while to do that as two patches, so I
hope it's OK that I just did it all as one patch.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A user reported that setting model=imac24 used to allow sound to work on their
Mac Pro 5,1 machine. Commit 5671087ffa "Move ALC885 macpro and imac24 models
to auto-parser" removed this model option. All Mac machines are now explicitly
handled with a quirk and the auto-parser. This adds a quirk for the device
found on the Mac Pro 5,1 machines.
This (partially) fixes https://bugzilla.redhat.com/show_bug.cgi?id=808559
[sorted the new entry in the ID number order by tiwai]
Reported-by: Gabriel Somlo <somlo@cmu.edu>
Signed-off-by: Josh Boyer <jwboyer@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's compatible with 8930G.
Using the same fixup gives the proper 5.1 sound back.
Reported-and-tested-by: Dany Martineau <dany.luc.martineau@gmail.com>
Cc: <stable@kernel.org> [v3.3+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add GPIO1 setup explicitly for Acer Aspire 493x & co.
This could be set by alc_auto_init_amp(), but it's safer to set it
more explicitly in the fixup table.
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>
Also remove two warnings when CONFIG_SND_DEBUG is not set:
sound/pci/hda/patch_hdmi.c: In function ‘hdmi_intrinsic_event’:
sound/pci/hda/patch_hdmi.c:761:6: warning: unused variable ‘eldv’ [-Wunused-variable]
sound/pci/hda/patch_hdmi.c:760:6: warning: unused variable ‘pd’ [-Wunused-variable]
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A bunch of driver-specific fixes and one generic fix for the new support
for platform DAPM contexts - we were picking the wrong default for the
idle_bias_off setting which was meaning we weren't actually achieving
any useful runtime PM on platform devices.
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Merge tag 'asoc-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: fixes for 3.4
A bunch of driver-specific fixes and one generic fix for the new support
for platform DAPM contexts - we were picking the wrong default for the
idle_bias_off setting which was meaning we weren't actually achieving
any useful runtime PM on platform devices.
name pins consistently (MIC1/LINE1/HP-OUT/CD) on all controls
affecting those pins.
remove duplicate SET_AMP_GAIN_MUTE to 0x17/index 0 and 0x17/index 1
really select MIC1, not Mixer out for recording
"Mixer out" for recording is not a "pin", adjust comment
Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CX20549 (ctx5045) doesn't accept data on index 1 for output pins,
as shown in the following hda-var transaction:
$ hda-verb /dev/snd/hwC0D0 0x10 set_amp_gain 0xb126
nid = 0x10, verb = 0x300, param = 0xb126
value = 0x0
$ hda-verb /dev/snd/hwC0D0 0x10 get_amp_gain 0x8001
nid = 0x10, verb = 0xb00, param = 0x8001
value = 0x0
Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ID used for detection of the BenQ R55E actually identifies the
Quanta TW3 ODM design, which is also used for the Gigabyte W551 laptop
series. Schematics on the internet clearly indicate that the "Port C"
(analog input connected to record source #4 and mixer input #4) is
unconnected.
Playing an audio CD through analog playback (using cdplay from cdtools)
produces no sound, even with the mixer input labelled "CD" enabled, and
the volume control in the CD drive set to maximum. This indicates the
connection is really not present.
Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The "input converter" widget of the CX20459 has only one input amplifier,
expose that one as "Capture Volume/Capture Switch". The actual record
source selection is already exposed through the separately installed
input mux.
Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This includes renaming "Line In" to line, also in the mixer settings.
Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The CX20549 has only one single input amp on it's input converter
widget. Fix printing of values in the codec file in /proc/asound.
Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A collection of small fixes for 3.4-rc1, including
- mic-recording regression fix for Realtek codec
- clean-up of dmaengine parameter mess
- WM8894 calibration tweak
- minor fixes for asihpi and some bool module parms
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of small fixes for 3.4-rc1, including
- mic-recording regression fix for Realtek codec
- clean-up of dmaengine parameter mess
- WM8894 calibration tweak
- minor fixes for asihpi and some bool module parms"
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: fix isa/opti9xx module param type
sound: fix oss/msnd_pinnacle module param type
ALSA: asihpi - fix return type of hpios_locked_mem_alloc()
ASoC: dmaengine_pcm: use dmaengine cyclic wrapper
ASoC: Add extra parameter to device_prep_dma_cyclic
ALSA: hda/realtek - Fix ADC assignment with a shared HP/Mic pin
ASoC: wm8994: Update WM8994 DCS calibration
Here is the first big update chunk of sound stuff for 3.4-rc1.
In the common sound infrastructure, there are a few changes for
dynamic PCM support (used in ASoC) and a few clean-ups. Majority of
changes are found, as usual, in HD-audio and ASoC.
Some highlights of HD-audio changes:
- All the long-standing static quirk codes for Realtek codec were
finally removed by fixing and extending the Realtek auto-parser.
- The mute-LED control is standardized over all HD-audio codec
drivers using the extended vmaster hook.
- The vmaster slave mixer elements are initialized to 0dB as default
so that the user won't be annoyed by the silent output after
updates, e.g. due to the additions of new elements.
- Other many fix-ups for the misc HD-audio devices.
In the ASoC side, this is a very active release, including a quite a
few framework enhancements. Some highlights:
- Support for widgets not associated with a CODEC, an important part
of the dynamic PCM framework.
- A library factoring out the common code shared by dmaengine based
DMA drivers contributed by Lars-Peter Clausen. This will save a lot
of code and make it much easier to deploy enhancements to
dmaengine.
- Support for binary controls, used for providing runtime
configuration of algorithm coefficients.
- A new DAPM widget type for regulator supplies allowing drivers for
devices that can power down unused supplies while active to do
without any per-driver code.
- DAPM widgets for DAIs, initially giving a speed boost for playback
startup and shutdown and also the basis for CODEC<->CODEC DAI link
support.
- Support for specifying the number of significant bits on audio
interfaces, useful for allowing applications to know how much effort
to put into generating data for a larger sample format.
- Conversion of the FSI driver used on some SH processors to
DMAEngine.
- Conversion of EP93xx drivers to DMAEngine.
- New CODEC drivers for Maxim MAX9768 and Wolfson Microelectronics
WM2200.
- Move audmux driver from arc/arm to sound/soc
- McBSP move from arch/ to sound/ and updates
Also, a few small updates and fixes for other drivers like au88x0,
ymfpci, USB 6fire, USB usx2yaudio are included.
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull updates of sound stuff from Takashi Iwai:
"Here is the first big update chunk of sound stuff for 3.4-rc1.
In the common sound infrastructure, there are a few changes for
dynamic PCM support (used in ASoC) and a few clean-ups. Majority of
changes are found, as usual, in HD-audio and ASoC.
Some highlights of HD-audio changes:
- All the long-standing static quirk codes for Realtek codec were
finally removed by fixing and extending the Realtek auto-parser.
- The mute-LED control is standardized over all HD-audio codec
drivers using the extended vmaster hook.
- The vmaster slave mixer elements are initialized to 0dB as default
so that the user won't be annoyed by the silent output after
updates, e.g. due to the additions of new elements.
- Other many fix-ups for the misc HD-audio devices.
In the ASoC side, this is a very active release, including a quite a
few framework enhancements. Some highlights:
- Support for widgets not associated with a CODEC, an important part
of the dynamic PCM framework.
- A library factoring out the common code shared by dmaengine based
DMA drivers contributed by Lars-Peter Clausen. This will save a
lot of code and make it much easier to deploy enhancements to
dmaengine.
- Support for binary controls, used for providing runtime
configuration of algorithm coefficients.
- A new DAPM widget type for regulator supplies allowing drivers for
devices that can power down unused supplies while active to do
without any per-driver code.
- DAPM widgets for DAIs, initially giving a speed boost for playback
startup and shutdown and also the basis for CODEC<->CODEC DAI link
support.
- Support for specifying the number of significant bits on audio
interfaces, useful for allowing applications to know how much
effort to put into generating data for a larger sample format.
- Conversion of the FSI driver used on some SH processors to
DMAEngine.
- Conversion of EP93xx drivers to DMAEngine.
- New CODEC drivers for Maxim MAX9768 and Wolfson Microelectronics
WM2200.
- Move audmux driver from arc/arm to sound/soc
- McBSP move from arch/ to sound/ and updates
Also, a few small updates and fixes for other drivers like au88x0,
ymfpci, USB 6fire, USB usx2yaudio are included."
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (446 commits)
ASoC: wm8994: Provide VMID mode control and fix default sequence
ASoC: wm8994: Add missing break in resume
ASoC: wm_hubs: Don't actively manage LINEOUT_VMID_BUF
ASoC: pxa-ssp: atomically set stream active masks
ASoC: fsl: p1022ds: tell the WM8776 codec driver that it's the master
ASoC: Samsung: Added to support mono recording
ALSA: hda - Fix build with CONFIG_PM=n
ALSA: au88x0 - Avoid possible Oops at unbinding
ALSA: usb-audio - Fix build error by consitification of rate list
ASoC: core: Fix obscure leak of runtime array
ALSA: pcm - Avoid GFP_ATOMIC in snd_pcm_link()
ALSA: pcm: Constify the list in snd_pcm_hw_constraint_list
ASoC: wm8996: Add 44.1kHz support
ALSA: hda - Fix build of patch_sigmatel.c without CONFIG_SND_HDA_POWER_SAVE
ASoC: mx27vis-aic32x4: Convert it to platform driver
ALSA: hda - fix printing of high HDMI sample rates
ALSA: ymfpci - Fix legacy registers on S3/S4 resume
ALSA: control - Fixe a trailing white space error
ALSA: hda - Add expose_enum_ctl flag to snd_hda_add_vmaster_hook()
ALSA: hda - Add "Mute-LED Mode" enum control
...
The recent Realtek driver tries to assign an extra input via the
headphone plug when only a single input source is found. The code
worked on Samsung Q1, but it broke ASUS 1015 where the mic is a
digital-mic and only a specific ADC works.
This patch fixes the assignment of ADC in the shared mic/hp case.
Instead of assuming the single ADC at first, reduce the ADCs after
trying to assign both mic and HP pins.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42973
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pull trivial tree from Jiri Kosina:
"It's indeed trivial -- mostly documentation updates and a bunch of
typo fixes from Masanari.
There are also several linux/version.h include removals from Jesper."
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial: (101 commits)
kcore: fix spelling in read_kcore() comment
constify struct pci_dev * in obvious cases
Revert "char: Fix typo in viotape.c"
init: fix wording error in mm_init comment
usb: gadget: Kconfig: fix typo for 'different'
Revert "power, max8998: Include linux/module.h just once in drivers/power/max8998_charger.c"
writeback: fix fn name in writeback_inodes_sb_nr_if_idle() comment header
writeback: fix typo in the writeback_control comment
Documentation: Fix multiple typo in Documentation
tpm_tis: fix tis_lock with respect to RCU
Revert "media: Fix typo in mixer_drv.c and hdmi_drv.c"
Doc: Update numastat.txt
qla4xxx: Add missing spaces to error messages
compiler.h: Fix typo
security: struct security_operations kerneldoc fix
Documentation: broken URL in libata.tmpl
Documentation: broken URL in filesystems.tmpl
mtd: simplify return logic in do_map_probe()
mm: fix comment typo of truncate_inode_pages_range
power: bq27x00: Fix typos in comment
...
A previous commit af65cbf296 (ALSA: hdmi: fix printout of SAD sampling
rates) fixed the sample rates shown in /proc/asound/cardX/eldY and
kernel log to not be entirely wrong. However, a missing rate from the
array added in the patch causes HDMI rates 88.2 kHz, 96 kHz, 176.4 kHz,
and 192 kHz to be shown as 96 kHz, 176.4 kHz, 192 kHz, and 384 kHz,
respectively.
Fix the reporting by adding the ALSA rate 64 kHz into the conversion
array between 48 kHz and 88.2 kHz.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since it's not always safe to assume that the vmaster hook is purely the
mute-LED control, add the flag indicating whether to expose the mute-LED
enum control or not. Currently, conexant codec sets this off for non-HP
laptops where EAPD may be used really as EAPD.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create snd_hda_add_vmaster_hook() and snd_hda_sync_vmaster_hook()
helper functions to handle the mute-LED in vmaster hook more
commonly. In the former function, a new enum control "Mute-LED Mode"
is added. This provides user to choose whether the mute-LED should be
turned on/off explicitly or to follow the master-mute status.
Reviewed-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the vmaster hook for controlling EAPD dynamically to Conexant
auto-parser. When the Master is muted, EAPDs are turned off as well.
This will fix the missing mute-LED control on some machines in
addition to the more power-saving in the auto-parser mode.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've had ugly static handling of the mute-LED with a powersave hook
for ALC269 HP laptops just like done in patch_sigmatel.c. This is now
rewritten with the new vmaster hook and a fixup code.
For that, the new fixup action, ALC_FIXUP_ACT_BUILD, is introduced.
It's called after build_controls is called. The reason of this new
action is that vmaster hook must be added at this stage (not in init
or probe).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The mute-LED is controlled in patch_sigmatel.c by (ab-)using the
powersave hook. This can be now rewritten with the vmaster hook
instead, which is much simpler and can work even without
CONFIG_SND_HDA_POWER_SAVE kconfig.
A drawback is that the mute-LED corresponds _only_ to the Master mixer
switch instead of checking the whole DACs. But usually this shouldn't
be a big problem as PA enables the mixer elements accordingly.
Also, this patch changes the code to create vmaster always even on
STAC9200 and STAC925x. The former "Master" on these chips are renamed
as "PCM" now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the driver is changed to use vmaster or a new slave element is
added by the improvement of the parser code, user may face often the
silent output because of the muted slave mixer although Master volume
is properly set. And they complain. And I get upset.
Although such a mixer element should be initialized via "alsactl init",
it'd be more user-friendly if the known output slaves are unmuted and
set to 0dB so that user can control the volume only with Master as
default. Since Master is still set muted as default even with this
change, no risk of the speaker blow up, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've seen a problem with a pop-noise at suspend/resume on a HP
machine with ALC269, and it turned out to be an issue that the
controller going to D3 while the codec is unused.
When the device is once suspended and resumed and kept unused, the
driver doesn't initialize the codecs. Instead, the codec chips are
set up dynamically at the first usage. Now, suppose the device going
to suspend again before the codec is set up. The controller is turned
off to D3 while the codec chips are untouched. This caused a pop
noise because the codec chip might have been turned on implicitly by
the hardware.
As a workaround, the codec chip needs to be set to D3 when going to
suspend no matter whether it was used or not. Also, for making it
happening, the controller has to be always set up in the resume path.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the currently unused spec->init_hook at the beginning of the init
sequence so that the recently added ALC269VB coef setup can be put
there. The alc_init() is again clean without an ugly check.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The coef setup in alc269_fill_coef() was designed only for ALC269VB
model, and this has some bad effects for other ALC269 variants, such
as turning off the external mic input. Apply it only to ALC269VB.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
My CD input got lost in commit 68ef0561ef.
Raymond helped me to add the necessary pin fixup to make it appear again. In
fact, this is basically his patch. It fixes alsa bug #5541.
Signed-off-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent addition of volume-knob widget in the auto-parser broke
automute/autoswitch for some Realtek devices.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It seems that Lenovo machines (or codec chip itself?) leave the unsol
event tags and the enablement-flag from other pins bogusly even on the
unused pins. Although this shouldn't be too critical, it's better to
clear them up sanely.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some codecs don't supply the mute amp-capabilities although the lowest
volume gives the mute. It'd be handy if the parser provides the mute
mixers in such a case.
This patch adds an extension amp-cap bit (which is used only in the
driver) to represent the min volume = mute state. Also modified the
amp cache code to support the fake mute feature when this bit is set
but the real mute bit is unset.
In addition, conexant cx5051 parser uses this new feature to implement
the missing mute controls.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42825
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A bug report with an old Sony laptop showed that we can't rely on BIOS
setting the pins of headphones but the driver should set always by
itself.
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The docking-station of Thinkpad X200 & co supports also an SPDIF output,
and the corresponding pin 0x1c has to be enabled for using it.
Reported-and-tested-by: Sebastian Glita <sebastian.glita@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
Enables port E of IDT 92HD91 codec as output and sets correct output
phase between ports E and D and high pass filter.
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When there are multiple input sources, the driver wrongly overwrites with
the value of the last input source on other slots at resume. Thus the
primary input source may be shown wrongly.
Reported-and-tested-by: Julian Sikorski <belegdol@gmail.com>
Cc: <stable@kernel.org> [v3.1+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
