Instead of fixed "Front" mixer name, try to assign a better name, e.g.
"Master" or "Speaker" fot the primary output volume controls of ALC260
codec.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When there is only one DAC is used for ALC880, try to assign a better
name, either Speaker or Front, depending on the output pin type.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Provide a standard parser for input pins to create the input mixer
and input source controls instead of having a difference one for each
Realtek codec. The new helper parses the codec connections dynamically
isntead of fixed indicies.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reuse a part of the code of ALC268 parser for ALC269.
This will change the default output volume either to Front or Speaker
depending on the pin configuration.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a logic error in the range check of the input level control that
would prevent setting any volume less than the maximum.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Modify loops in such way that the register value is checked also after
the timeout condition, just in case the heavy interrupt load etc. caused
the thread to sleep for the time period exceeding the timeout value.
While at it remove an extra ALI_STIMER read from snd_ali_stimer_ready().
Reported-by: Jack Byer <ojbyer@usa.net>
Signed-off-by: Bartlomiej Zolnierkiewicz <bzolnier@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are many variants of Toshiba laptops with ALC268 codec, and
it seems that a few of them don't work with model=toshiba preset
since they have the secondary ALC268 codec just for HDMI output.
This is a regression due to the previous clean-up work to merge all
Toshiba quirk entries into a single check.
This patch adds the identification of such laptops to apply the
standard BIOS-probing method. Unfortunately, Toshiba laptops have
all the same PCI SSID, so we need to check the codec SSID to identify
each device.
Tested-by: Alexey Dobriyan <adobriyan@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The BIOS pin configs are in fact correct and shall not be overwritten.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HP Compaq 6530s and 6531s internal speaker is silence or becomes silence
within 1 minute after fresh boot. It is found that pin 0x1c must be set to
PIN_OUT mode to make the speaker work. This is weird - line-in pin 0x1c and
speaker pin 0x16 seem to be unrelated.
The codec differences before/after patch are:
@@ Node 0x17 [Pin Complex] wcaps 0x40020b:
Pin Default 0x41a6e130: [N/A] Mic at Ext Rear
Conn = Digital, Color = White
DefAssociation = 0x3, Sequence = 0x0
Misc = NO_PRESENCE
- Pin-ctls: 0x24: IN
+ Pin-ctls: 0x40: OUT
@@ Node 0x1c [Pin Complex] wcaps 0x40018d:
Pin Default 0x41813021: [N/A] Line In at Ext Rear
Conn = 1/8, Color = Blue
DefAssociation = 0x2, Sequence = 0x1
- Pin-ctls: 0x24: IN VREF_80
+ Pin-ctls: 0x40: OUT VREF_HIZ
Unsolicited: tag=00, enabled=0
Connection: 1
0x24
Tests show that it won't impact (external) Mic recording.
Reported-by: "Lin, Ming M" <ming.m.lin@intel.com>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The auto-mic clean-up patches caused regressions on some ALC268 models
that have no proper input_mux but with "Input Source" mixer elements.
Such a combination results in Oops when accessed.
[A reason why set_capture_mixer() isn't used in patch_alc268() is that
ALC268 codec have HDA_OUTPUT direction for capture volumes unlike other
codecs. Thus it needs own definitions of capture elements.]
This patch fixes the issues:
- Add a capture mixer definition without input-source
- Use the new capture mixer appropriately
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The previous fix removed the definition of num_adc_nids wrongly, and
this resulted in the missing input-source control. Now readded again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few improvements for IDT 92HD83xxx codec pareser:
- Remove unused / deprecated mixer-amp controls
- Handle d-mics as normal inputs since this codec has no separate
MUXes for analog and digital
- Don't create duplicated controls for capture volumes with Mux
capture volumes
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Enable line-out detection for IDT/STAC codecs only when speaker pins
exist. In some cases, the speaker itself is identified as line-out,
and this confuses the situation. Only the extra line-outs should do
auto-muting.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
the hdsp driver refuses to report any information via the proc
interface, if the io box is not connected. with this patch, the
content of the control and status registers is printed before the
iobox check.
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With auto config model of alc268 realtek codec, it allows to select any
of possible available digital microphone inputs when only one is
available. For example, when only digital mic in nid 0x12 is available,
on second input source it will allow you to select unavailable digital
mic in nid 0x13. The problem is that selecting unavailable digital mic
creates a source of noise when recording (I'm not sure if this happens
on all machines with alc268 and only one digital mic input, but testing
on a quanta uw1 netbook a lot of noise is introduced in recording from
digital mic 0x12/first input source, when you select the unavailable
digital mic 0x13 for capture source 0x24 in the second input source in
mixer).
Then to avoid noise when recording from digital mic with auto model in
this case, prevent a digital mic input source to be selected if
microphone is not available.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move static codes to setup from init_hook for each model.
Also, use the common auto-mic selection helper for devices that support
auto-mic selection. They just need to set up ext_mic, int_mic and
auto_mic flag in the setup section.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added setup hook to ALC preset struct to be called at in the parser
but not at each init callback.
This can be used for setting up the static pins, etc, while the
init hook should be used for updating the status again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Created a white-list to enable MSI since some devices require MSI
explicitly due to BIOS/ACPI problems. Simply using a quirk list.
As the first case, take HP Compaq CQ40.
Reference: Novell bnc#529971
https://bugzilla.novell.com/show_bug.cgi?id=529971
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some Realtek codecs don't provide the full connections for certain pins
from each ADC; e.g. ACL662/ALC272 gives only one of two digital-mic pins
for each ADC. Thus, depending on the digital mic pin, the ADC/MUX to be
used has to be chosen properly.
This patch adds the check of the connectivity of pins at auto-mic mode.
If no proper connectivity is found, auto_mic flag is turned off to be
sure.
Also the mux_idx is determined during this check so it won't be checked
in the unsol event any more.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC269 and ALC861-VD parsers override the ADC definitions
unconditionally without checking the spec definition. This causes
the problem when any inconsistent ADC is set up in the device quirk
(like ALC272 with digital-mic).
This patch avoids the overriding by adding the proper checks.
Reference: Novell bnc#529467
https://bugzilla.novell.com/show_bug.cgi?id=529467
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the support for automatic mic selection via plugging for
Realtek codecs (in auto-probing mode). The auto-mic mode is enabled
only when one internal mic and one external mic are present.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow the interval timer to be programmed with its full 96 kHz
precision.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Without the initialization of vmaster NID, the dB information got
confused for ALC269 codec.
Reference: Novell bnc#527361
https://bugzilla.novell.com/show_bug.cgi?id=527361
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
The previous auto-mic patch for STAC/IDT codecs causes the Oops on
machines without digital mic pins. This patch fixes the problem.
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 - Read buffer overflow
ALSA: hda: Correct EAPD for Dell Inspiron 1525
ALSA: hda: warn on spurious response
ALSA: hda: remember last command for each codec
ALSA: hda: read CORBWP inside reg_lock
ALSA: hda: take reg_lock in azx_init_cmd_io/azx_free_cmd_io
ALSA: hda: take cmd_mutex in probe_codec()
ALSA: hda: track CIRB/CORB command/response states for each codec
ALSA: hda - Fix quirk for Toshiba Satellite A135-S4527
When the line-out jack is plugged/unplugged, the driver needs to check
the headphone plug, not only the line-out jack itself. Otherwise the
headphone or the speaker may be wrongly muted/unmuted.
As a result, both STAC_HP_EVENT and STAC_LO_EVENT need to call the
same function, stac92xx_hp_detect().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit fefd67f31e
ALSA: hda - Add line-out jack detection on IDT/STAC codecs
enabled wrong pins for jack detections. Fixed to the correct ones.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The new IbexPeak HDMI codec has 3 pin nodes and 2 converter nodes.
Here we assume only the first ones will be used.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check whether index is within bounds before testing the element.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 24918b61b5 statically changes
the model from dell-bios to dell-3stack to solve the sound decreasing
regression (http://lkml.org/lkml/2008/9/12/203), however it leads to another
problem that the 2nd headphone jack doesn't work
(https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3987). So I think
the commit 249**2dc is just a workaround. I would like to give a true solution
here.
The datasheet for STAC9228 says, GPIO2 is the same pin as VOL DOWN, and
the EAPD pin is GPIO0. This is why the sound decreases if we set EAPD as
GPIO2. This patch changes EAPD to GPIO0 to solve the problem.
Signed-off-by: Chengu Wang <wangchengu@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This converts the last CORBWP access outside of reg_lock.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just for safety. azx_init_cmd_io() and azx_free_cmd_io() may be
called when switching to single command mode.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that each codec will have its own module, it is possible
for the user to load one codec while another one is running.
So cmd_mutex would be a safe addition to probe_codec().
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Recently we hit a bug in our dev board, whose HDMI codec#3 may emit
redundant/spurious responses, which were then taken as responses to
command for another onboard Realtek codec#2, and mess up both codecs.
Extend the azx_rb.cmds and azx_rb.res to array and track each codec's
commands/responses separately. This helps keep good codec safe from
broken ones.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use model=lenovo instead of model=dallas for Toshiba Satellite A135-S4527
with ALC861-VD codec.
Reference: Novell bnc#526325
https://bugzilla.novell.com/show_bug.cgi?id=526325
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The name buf with size 16 is too short for some codec names, e.g.
truncated like "ALC861-VD Analo". Now the size is doubled.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the automatic mute of speakers via line-out jack plugging on
STAC/IDT codecs. The feature is enabled when the HP detect is present.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch below, to be applied on the latest sound-unstable-2.6.git,
enables headphones output on my MacBookPro 5,5, together with the
automuting feature.
Here is the exact soundcard id:
Vendor Id: 0x10134206
Subsystem Id: 0x106b4d00
Revision Id: 0x100301
Signed-off-by: Takashi Iwai <tiwai@suse.de>
STAC/IDT codecs provide both "Input Source" and "Digital Input Source"
controls to choose the analog input source and the digital input source.
But this is far user-unfriendly.
This patch merges the input source selections into one "Input Source"
control. To have separate digital and analog input source controls,
you can pass "separate_dmux = 1 " hint string.
At the same time, this patch gets rid of analog mixer stuff that was
already disabled in previous patches.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It auto mutes all 8-channel outputs at rear panel when
the front panel headphone is connected.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This 2-channel mode is useful in that it will broadcast
a 2-channel audio stream to all front/side/... ports.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The analog mix is disabled now as default (unless "analog_mixer" hint
is given), so it shoudn't appear in the digital input source as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the missing initialization of DMUX connection (to analog input)
for auto-mic mode with STAC/IDT codecs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We don't need any more static connection to the port F (which is often
used for docking stations) since its connection is done dynamically via
DAC assignment now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
e->sad[] is declared with size ELD_MAX_SAD=16, but the guard
allows range 0-31.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Support the automatic mic-switching with some devices with IDT/STAC
codecs. The condition is that the device has only two inputs, one
for an external mic and one for an internal mic.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since only one event can be associated to a (pin) widget, it's safer
to avoid the multiple mapping. This patch fixes the behavior of the
STAC/IDT codec driver.
Now stac_get_event() doesn't take the type argument but simply returns
the first hit element. Then enable_pin_detect() checks the validity
of the type, and returns non-zero only if a valid entry. The caller
can call stac_issue_unsol_event() after checking the return value.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The analog mixer unit on IDT 92HD71Bxx codecs is almost useless
since we use only the direct connections from DAC to pin.
Remove the controls to avoid unneeded confusion as default now.
This can be still back via "analog_mixer = 1" hint.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of static snd_kcontrol_new arrays, create "Capture Volume"
and "Capture Switch" controls dynamically based on the mixer attr
values (made via HDA_COMPOSE_AMP_VAL()).
This reduces the code size and gives more flexibility to change
the number of controls later.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current driver creates always the digital input source mixer
elements for IDT 92HD71x codecs no matter whether digital mics are
present. This patch adds the proper check to avoid the creation of
these controls if unnecessary.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The sentense "Unknown model for xxx, ..." makes people too nervous
and drives them to a direction to a wrong "fix" by giving any
mismatching model option.
Let's rephrase the messages to be more nice and easy (at least that
won't make people suspect conspiracies).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Volume-knob widgets may have connections even if they have no CONN_LIST
cap bit. Allow the query exceptionally in snd_hda_get_connections().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* fix/hda:
ALSA: hda - Fix mute control with some ALC262 models
ALSA: hda - Restore GPIO1 properly at resume with AD1984A
ALSA: hda - Use snprintf() to be safer
The master mute switch is wrongly implemented as checking the pointer
instead of its value, thus it can be never muted. This patch fixes
the issue.
Reference: Novell bnc#404873
https://bugzilla.novell.com/show_bug.cgi?id=404873
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
Here are the new sound enabling patches for IbexPeak.
Summary of tested features:
- playback
- Front Headphone: OK
- 8 channel audio: Front/Rear/CLFE/Side all OK
- recording
- Front Mic/Rear Mic: both OK
(front/rear/line mics are selectable in the "Input source" alsamixer control)
- Line In: not working
(in 6ch mode, its amp/mute, direction and route all looks fine,
so I'm a little puzzled)
(hopefully no one will care this feature)
- digital SPDIF input/output: not tested (no equipment)
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 099db17e66 introduced a
regression at suspend/resume where the GPIO1 bit isn't properly
restored, thus the speaker output gets muted initially after resume.
The fix is simple, use the cached write for storing GPIO data.
Reference: Novell bnc#522764
https://bugzilla.novell.com/show_bug.cgi?id=522764
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a few uninitialized error checks that were introduced recently
mistakenlly during the clean-up:
sound/pci/ctxfi/ctamixer.c: In function ‘get_amixer_rsc’:
sound/pci/ctxfi/ctamixer.c:261: warning: ‘err’ may be used uninitialized in this function
sound/pci/ctxfi/ctamixer.c: In function ‘get_sum_rsc’:
sound/pci/ctxfi/ctamixer.c:415: warning: ‘err’ may be used uninitialized in this function
sound/pci/ctxfi/ctsrc.c: In function ‘get_srcimp_rsc’:
sound/pci/ctxfi/ctsrc.c:742: warning: ‘err’ may be used uninitialized in this function
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a check to snd_hda_get_connections() routine for
presence of AC_WCAP_CONN_LIST. Also, make sure that negative error
codes from noted route are handled on all places as errors.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Previous patch used widget type, but the presence flag of the connection
list is in the widget capabilities.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reading node connections for an unknown widget can confuse HDA codec bus.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the logic of ALC861 auto-mode parser for the outputs.
Instead of assuming the fixed DAC list, parse the conection and assign
the DAC dynamically.
Also, unmute the unused output connections to avoid noises on inputs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add some tricks to reduce the click noise at powering down to D3
in the power saving mode on STAC/IDT codecs.
The key seems to be to reset PINs before the power-down, and some
delay before entering D3. The needed delay is significantly long,
but I don't know why.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* fix/misc:
ALSA: ca0106 - Fix the max capture buffer size
ALSA: OSS sequencer should be initialized after snd_seq_system_client_init
ALSA: sound/isa: convert nested spin_lock_irqsave to spin_lock
* fix/hda:
ALSA: hda - Fix pin-setup for Sony VAIO with STAC9872 codecs
ALSA: hda - Add quirk for Gateway T6834c laptop
ALSA: hda_codec: Check for invalid zero connections
On Soundblaster X-FI Titanium with emu20k2 the SIDE and SURROUND
channels were swapped and wrong.
I double checked it with connector colors and creative soundblaster
windows drivers.
So I swapped them to the true order.
Now "speaker-test -c6" and "speaker-test -c8" are working fine.
Signed-off-by: Frank Roth <frashman@freenet.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The capture buffer size with 64kB seems broken with CA0106.
At least, either the update timing or the DMA position is wrong,
and this screws up pulseaudio badly.
This patch restricts the max buffer size less than that to make life
a bit easier.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
The recent rewrite of the codec parser for STAC9872 caused a regression
for some Sony VAIO models that don't give proper pin default configs
by BIOS. Even using model=vaio doesn't work because the pin definitions
are set after the pin overrides.
This patch fixes the pin definitions in patch_stac9872() to be put
in the right place before the pin overrides. Also the patch adds the
new quirk entry for VAIO F/S to have the correct pin default configs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
Added the native timer support for emu20k2, which gives much more
accurate update timing than the system timer.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Gateway T6834c laptops need EAPD always on while the default behavior
for the STAC9205 reference board is to turn it off upon every HP plug.
By using the special "eapd" model, which is first introduced for Gateway
T1616 laptops for this same reason, this peculiarity can be properly
handled.
Signed-off-by: Hao Song <baritono.tux@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To prevent "Too many connections" message and the error path for some HDMI
codecs (which makes onboard audio unusable), check for invalid zero
connections for CONNECT_LIST verb.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Parse the mono output pin 0x16 correctly even as the primary output
- Create "Speaker" volume control if the primary output is a speaker
- Fix the wrong direction of (optional) "Mono" switch
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The newly added sanity-check for a codec verb can be better written
with logical ORs. Also, the parameter can be more than 8bit.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
