In patch_vt1708(), the check of MUX nids is missing and this results in
the -EINVAL error in accessing Input Source mixer element. Simpliy
adding the call of get_mux_nids() fixes the problem.
Reference: Novell bnc#534904
https://bugzilla.novell.com/show_bug.cgi?id=534904
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, the digital mic capture volume wasn't created. This is because
IDT codecs have output amps for digital mics, not input amps, while
input amps should be used for other analog pins. Thus the automatic
capture volume creation should check both directions for digital mics.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
1) Added support of internal subwoofer (it sounds!!!)
2) Auto muting front speakers and internal subwoofer on headphones plug.
3) Internal mic works.
4) 3 channel mods (jack maps):
black pink blue
2ch: front ext mic line in
4ch: front ext mic surround
6ch: front CLFE surround
Can be changed in mixer.
5) Sound can be recorded from:
Internal mic
Ext mic
Cd
Line in
6) 2 separate capture channels.
Signed-off-by: Denis Kuplyakov <dener.kup@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
One more patch to give a better name for the primary output controls,
this time for ALC861-VD codec. The change is simple, just checking the
pin connection whether it's a speaker-out. When both speaker and HP
are assigned, we name the volume as "PCM" as this influences on both
outputs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Similar improvements for ALC262 codec like previous two commits:
assign a better name, either Master or Speaker, for the primary output
controls.
However, in the case of ALC262 codec, the necessary changes are larger
than others because we need to check the possibility of different mixer
amps depending on the pins. The pin 0x16 is mono, and bound with the
dedicated mixer 0x0e while other pins are bound with 0x0c. Thus, there
are two possible volumes.
When only one of them is used, we can name it as "Master". OTOH, when
both are used at the same time, they have to be named uniquely.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
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>
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>
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>
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>
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>
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>
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>
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>
A recent bug involves passing auto detected >0x7f NID to codec command,
creating an invalid codec addr field, and finally lead to cmd timeout
and fall back into single command mode. Jaroslav fixed that bug in
alc880_parse_auto_config().
It would be safer to further check the bounds of all cmd fields.
Cc: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for new AMD HD audio devices. Use generic driver to detect HD audio
devices with Vendor ID AMD.
Signed-off-by: Andiry Xu <andiry.xu@amd.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds support for the Conexant CX20582 codec, based on code from
http://www.linuxant.com/alsa-driver/alsa-driver-linuxant-1.0.19ppch12-1.noarch.rpm.zip
This is the codec to be shipped in the OLPC XO-1.5, so this patch also
includes an XO-specific profile. Resultant configuration:
http://dev.laptop.org/~dsd/20090713/codec0.txthttp://dev.laptop.org/~dsd/20090713/codec0.svg
As the Linuxant code is structured differently to the other codecs,
I was unable to cleanly reimplement everything in the generic and Dell
profiles as more info is needed (e.g. codec graphs). I simplified those
profiles so that hopefully it will not break anyone's audio. If it does,
it may be worth returning -ENODEV from patch_cx5066 on non-OLPC systems,
and then fixing snd_hda_codec_configure() to fall back on the generic
parser, at least until support for other systems is figured out.
Signed-off-by: Daniel Drake <dsd@laptop.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The codec read errors in snd_hda_get_connections() are ignored so far,
and it causes a problem like the bug in the commit
9d30937acc
ALSA: hda_intel: more strict alc880_parse_auto_config dig_nid checking
Better to check errors in the function and returns a proper error code
rather than passing bogus NID values.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On some IbexPeak systems with ALC889A errors like "azx_get_response
timeout, switching to polling mode: last cmd=0xaf9f000b" are produced,
because non-existent codec #10 is wrongly accessed.
The problem is that snd_hda_get_connections() returns out-of-range result
for NID 0x1c (something like 0xf8f9 or 0xffff).
This patch adds a check to alc880_parse_auto_config() to avoid using
of this out-of-range NIDs. A better fix maybe to improve
snd_hda_get_connections() routine to check for valid NID ranges if
NIDs are expected as result.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the merge error at the commit 305355aad8,
an addition of the missing alc880_gpio3_init_verbs to ALC882_TARGA model.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Simplify ALC882_TARGA and return gpio3 to ALC883_TARGA_DIG and
ALC883_TARGA_2ch_DIG, which I accidentally removed in commit id
64a8be7435
Signed-off-by: David Heidelberger <d.okias@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the beep tone calculation for IDT/STAC codecs, lower numbers correspond
to higher frequencies and vice versa. The current code has this backwards,
resulting in beep frequencies which are way too high (and sound bad on
tinny laptop speakers, resulting in complaints).
[Also added hz <= 0 check by tiwai]
Signed-off-by: Paul Vojta <vojta@math.berkeley.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is a regression, introduced in aa202455ee
(in alsa-kernel) which I noticed when trying to use the headphone socket on
my EeeCPC 901: the output was *very* quiet, practically silent.
This patch corrects the control types to that which was obviously intended in
the referenced commit.
Signed-off-by: Darren Salt <linux@youmustbejoking.demon.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HDA driver disabled HD audio 64bit address support for all AMD
SB600/SB700/SB800 platforms with commit
09240cf429 due to one SB600 issue
reported by community, but we do not see the similar issue on
SB700/SB800 platforms.
This patch is to refine the workaround for SB600 only.
Signed-off-by: Andiry Xu <andiry.xu@amd.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check the widget type and don't take invalid widgets while parsing
the capture source in patch_via.c.
Also, fixed some compile warnings introduced in the previous commit.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The fixed widget NIDs in patch_via.c seem wrong for some codecs,
and it resulted in the invalid capture source selection.
This patch adds the code to parse the topology instead of using
fixed numbers in order to get the right MUX widget id corresponding
to the ADCs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the output pin is used and EAPD capability is present, turn on
the EAPD bit. This fixes the silent output problem on ASUS laptops
with VT1708S codec.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The parser wasn't called in the proper order.
Split now the parser to be called in patch_cirrus(), and the rest
are just for building PCMs and controls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When FLOAT PCM format is available but together with other linear
PCM formats, don't override maxbps value. For FLOAT format, it's always
32, thus it can be better checked in snd_hda_calc_stream_format().
Otherwise the maxbps 32 might be used wrongly even if the linear PCM
doesn't support it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add some sanity checks of struct snd_pcm_hardware fields in the PCM
open callback of hda driver. This makes a bit easier to debug any PCM
setup errors in the codec side.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM rates bit field may have been changed by the codec open callback.
In that case, we need to reset rate_min and rate_max. So, simply call
snd_pcm_lib_hw_rates() again after the codec open callback.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check whether formats and rates don't result in zero due to the
restriction of SPDIF sharing. If any of them can be zero, disable
the SPDIF sharing mode instead. Otherwise it will lead to a PCM
configuration error.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The mixer elements created for ASUS eeePC 1000 with ALC269 aren't
standard but strange words like "LineOut". Rename the element names
to follow the standard one like "Headphone" and "Speaker".
Also, split the volumes to each so that the virtual master can control
them.
The alc269_fujitsu_mixer is removed because it's now identical with
the new eeepc mixer.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HP laptops with AD1984A codecs (at least mobile models) need to set
GPIO1 appropriately to indicate the mute state. The BIOS checks this
bit to judge whether the mute on or off is sent via F8 key.
Without changing this bit, the BIOS can be confused and may toggle
the mute wrongly.
Reference: Novell bnc#515266
https://bugzilla.novell.com/show_bug.cgi?id=515266
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of expanding alc882_init_verbs to two elements via a macro,
manually expand to each entry. This makes clear that some have already
the full slot for init_verbs array (currently 5).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After merting patch_alc882() and patch_alc883(), the initialization of
mixer amp 0x0b was missing in alc882_base_init_verbs[].
This is usually no critical problem, but it can disable the power-saving
as the default state, so better to put to mute these channels.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, the FLOAT PCM format is used only exclusivley set. But
this can be a combination with other formats.
This patch changes the parser to allow the FLOAT format in addition
to other PCM formats.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
alc882_auto_init_analog_input() sets the input pins to VREF-80 regardless
of the input pin types although it shouldn't be for line-in pins.
This patch fixes the behavior to follow other codecs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge patch_alc882() and patch_alc883() to the former one since both
codecs have fairly similar connections but just a slight difference.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
During the changes to clean up / fix the realtek codec initialization
routines in commit 4a79ba34ca,
I forgot to add the check for ALC268 and ALC269.
This resulted in the missing EAPD and COEF setup for these codecs.
This patch adds the missing checks for these codecs.
Reference: bko#13633
http://bugzilla.kernel.org/show_bug.cgi?id=13633
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check the hint "beep" in snd_hda_attach_beep_device() to avoid the beep
device creation if user doesn't want.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Line In connector is set up as PIN_IN by default, using
VREF_HIZ. It is connected to both ADCs, so add it to both
input selectors.
Also add the ability to use the input mix (on a SoundBlaster
one would call this "What You Hear").
Signed-off-by: Tony Vroon <tony@linx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the following bugs of acer-aspire-6530g model with ALC888:
- HP jack to mute all speaker outputs including LFE
- Make digital built-in mic working
Signed-off-by: Emilio López <buhitoescolar@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Realtek codecs require the pin-sense trigger call before actually
reading the pin-sense. Without this, the pin-detection might not be
done accurately.
This patch adds the pin-capability check and issues the trigger call
if required.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Samsung P50 requires the HP auto-muting unlike other Samsung models.
Added a new model=samsung-p50 to support this.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new flag to ad_spec struct so that the same hack can be used for
any other models (if any). This also allows other models to reuse the
auto-mute functions.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Split mixer element arrays of AD1986A models to several pieces so that
each model can share the same mixer arrays.
This removes lots of duplicated data.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the patch module option to apply a "patch" as a firmware to
modify pin configurations or give additional hints to the driver
before actually initializing and configuring the codec.
This can be used as a workaround when the BIOS doesn't give sufficient
information or give wrong information that doesn't match with the real
hardware setup, until it's fixed statically in the driver via a quirk.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The codec setup call via snd_hda_codec_configure() isn't necessarily
called in snd_hda_codec_new(). For the later added feature, it's better
to change the code flow like:
- create all codec instances
- configure each codec
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make the jack-plug notification via input layer selectable via Kconfig.
This is often unnecessary, and the similr function will be provided
using the ALSA control API in near future anyway.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the digital-mic support with ALC262 auto model.
The new ALC262 models have the digital mic at NID 0x12. This widget
isn't checked in the current alc262_auto_create_analog_input_ctls()
since it's under 0x18. So, just reuse the routine for alc269 to fix
the behavior.
But, it doesn't suffice: the digital mic is supported only with the
ADC0, we have to exclude other ADCs when d-mic is detected.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the check of the input-source type by checking the widget type of
each capture-source item. Since some codecs can have both the mixer
and selector types depending on the ADC, alc_mux_enum_put() needs to
check each widget.
With this change, spec->capture_style gets unneeded, so it's removed,
too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit f9e336f65b
ALSA: hda - Unify capture mixer creation in realtek codes
removed the "Input Source" mixer element creation for toshiba-s06 model
because it contains a digital-mic input.
This patch take the control back.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
Fix the comparison of unsigned int that causes a compile warning below
by changing to the right signed type:
patch_sigmatel.c: In function ‘stac92xx_vref_set’:
patch_sigmatel.c:658: warning: comparison of unsigned expression < 0 is always false
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The selected 4930G model seemed to keep the subwoofer 'tuba'
function from operating correctly. Removing the existing PCI
ID match made this work again, but it was mapped to 'Side'
instead of to LFE as one would expect.
This attempts to enable all functionality and keep the amount
of available mixer sliders low. Any slider that had no audible
effect on the output audio has been removed, and as such EAPD
is not currently enabled.
Signed-off-by: Tony Vroon <tony@linx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Correct some cut+paste typos from 'tagra' to 'targa'.
Signed-off-by: Sasha Alexandr <brina_keith@ns.sympatico.ca>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added 7.1 support for MSI GX620 and jack quirk.
Reference: kernel bug#13430
http://bugzilla.kernel.org/show_bug.cgi?id=13430
Signed-off-by: David Heidelberger <d.okias@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
with BIOS probing only we offer a non functional headphone swith and
volume slider.
Signed-off-by: Guido Günther <agx@sigxcpu.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix this build error when CONFIG_PM is not set:
ound/pci/hda/hda_intel.c: In function 'azx_bus_reset':
sound/pci/hda/hda_intel.c:1270: error: implicit declaration of function 'snd_pcm_suspend_all'
sound/pci/hda/hda_intel.c:1271: error: implicit declaration of function 'snd_hda_suspend'
sound/pci/hda/hda_intel.c:1272: error: implicit declaration of function 'snd_hda_resume'
Signed-off-by: Alexander Beregalov <a.beregalov@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Enable all three capture channels, including the missing nid 7 which is
the only one capable of capturing DMIC input
Enable Headphone amp for the HP jack. This causes a volume boost for
headphones, but does not cause any noticeable effect for light loads
like other amps, so there is no need to make it configurable.
Add Input Mix capture mux setting to capture the output of the playback
input mux (that is, what goes out the speakers except for PCM)
Hack another coef register because the stereo DMIC for some reason
produces a nonstandard sum/difference signal. I found a bit to make it
just use the sum signal for both channels, which makes it behave like a
standard mono microphone. The stereo is useless anyway (they're 1cm apart).
Tested working: Three capture channels, mic in, line in, DMIC.
Tested not working: CD. Not sure why, might be unconnected in the actual
hardware or a CD drive issue.
Also looked at SPDIF. It appears to work (emitter lights up inside the
HP out jack) but I lack a proper miniTOSLINK cable to test it.
Signed-off-by: Hector Martin <hector@marcansoft.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Short story: this laptop has 5.1 built-in speakers which you *really*
want to use (the not-so-"sub" woofer is what makes the audio above
average for a laptop), so 6-channel support is important (plus a decent
asound.conf to upmix stereo). It also has the 3 typical jacks that ought
to have a selectable mode. And it's based on ALC889, which sucks.
Rationale/explanations:
The const_channel_count stuff was added because, for a laptop like this,
you always have 6 channels available (internal speakers) but still need
to set the mode for the 3 external jacks. Therefore, the device always
needs to be in 6-channel mode but there still needs to be a mixer
control for the jack mode. You could use line/mic-in at the same time as
the 6 internal speakers, for example. You might be tempted to make it
even smarter by dynamically switching the max channel count when
headphones are plugged in (therefore muting the internal speakers and
reducing the physical channel count to the jack channel mode), but as a
user I consider this to be harmful because I want the audio to blow up
to 6 channels / upmixed as soon as I unplug the headphones, and having
opened the device while in 2-channel mode would prevent this from
working (and always making 6-channel mode available doesn't do any harm).
The hardware needs EAPD turned on and the DACs routed to the internal
speaker pins, so the patch adds those verbs.
The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT work
by default, at least here. I wasted much time trying to talk to
Realtek/pshou about this, but they just kept sending me useless updates
to patch_realtek.c that did nothing relevant. In the end I gave up and
brute forced the issue by trying to flip every bit in the proprietary
coefficient registers, and eventually found the two magic registers that
need to be cleared to enable all DACs. I have only heard Acer users
complain, but that might be because ALC889 is pretty new and using 5.1
(and noticing the missing center/lfe channels) might not be that common.
If this is a generalized issue with all ALC889 systems then those verbs
should probably be moved to a common verb array.
The internal mic is untested and probably doesn't work.
These settings will probably work for other Acer Gemstone laptops with
the same 5.1 speaker config. When identified, those should be added to
the PCI subsystem ID list.
Signed-off-by: Hector Martin <hector@marcansoft.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The reset of a BUS controller during operations is somehow risky and
shouldn't be done inevitably for devices that have apparently no such
codec-communication problems.
This patch adds the check of the hardware and limits the bus-reset
capability.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some machines machine cause a severe CORB/RIRB stall in certain
weird conditions, such as PA access at the start up together with
fglrx driver. This seems unable to be recovered without the controller
reset.
This patch allows the bus controller reset at critical errors so
that the communication gets recovered again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, LG R510 is only able to produce sound on headphones, the
internal speakers are not working.
The user tested and confirmed that with model=Dell headphones,
internal speakers and the microphone are working flawlessly.
Tested-by: Serdar Soytetir <tulliana@gmail.com>
Signed-off-by: Ozan Çağlayan <ozan@pardus.org.tr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
this is a patch against current snapshot that adds:
6 channels support for the MB5 model
Signed-off-by: Kacper Szczesniak <kacper@qwe.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix issues for 3 generations of HP workstations.
The modest modifications do the following:
1. Change the second MIC from device 3 to device 1
2. Init the "boost" values to "0" by default
From: John Brown <john.brown3@hp.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch changes Line In as Out Switch and Mic In as Out Switch to
enums for consistency, and causes all mic and line in ports to be probed
and controls to be added appropriately.
Signed-off-by: Nickolas Lloyd <ultrageek.lloyd@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ICH6_GCTL_RESET was wrongly set to another bit by the commit
b21fadb9c1. This caused a problem when
the codec needs really a reset (e.g. recovering from the communication
error at probe).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added some missing register bits definitions to reduce magic numbers.
Also renamed some to follow the names on the datasheet.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the single_cmd mode, the hardware cannot store the multiple replies
like on RIRB, thus each verb has to sync and wait for the response no
matter whether the return value is needed or not. Otherwise it may
result in a wrong return value from the previous verb.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A docking mic control is shown by default. The Compaq Presario
CQ60 laptop has no docking connector, so designate it as a
CXT5051_HP model.
This makes the phantom mixer slider disappear.
Signed-off-by: Tony Vroon <tony@linx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a debug mode to make the codec communication
synchronous. Define SND_HDA_SUPPORT_SYNC_WRITE in hda_codec.c,
and the call of snd_hda_codec_write*() will become synchronous,
i.e. wait for the reply from the codec at each time issuing a verb.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the single_cmd mode, the current driver code doesn't do any update
for RIRB just for any safety reason. But, actually the RIRB and
single_cmd mode don't conflict. Unsolicited events can be delivered
even while using the single_cmd mode.
This patch allows the handling of unsolicited events with single_cmd
mode, just always checking RIRB independent from single_cmd flag.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a codec communication error occurs, the CORB/RIRB counters should
be reset first before re-issuing the verb. Simply call azx_free_cmd_io()
and azx_init_cmd_io() to achieve that.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some Realtek codecs like ALC861 seem to support only VREF50 while the
current driver assumes it's only VREF80. Check other VREF bits to set
the correct value.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent fix for the headphone volume control on IDT/STAC codecs
resulted in the removal of invalid "Side" volume eventually. But,
if the front panel doesn't exist, this setup could be regarded as a
sort of regression, as reported in kernel bug #13250.
Now as a workaround, a new model 5stack-no-fp is added so that the user
without the front panel can choose this one explicitly.
Reference: bko#13250
http://bugzilla.kernel.org/show_bug.cgi?id=13250
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASUS W5Fm needs the fixed codec-slots to probe to override the BIOS
problem like W5F.
Tested-by: Alp Kılıç <kilic.alp@gmail.com>
Signed-off-by: Ozan Çağlayan <ozan@pardus.org.tr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The digital beep tone is calculated in two different ways depending
on the codec chip. The standard one is using a divider, and another
one is a linear tone for IDT/STAC codecs. Currently, only the
latter type is used for all codecs, which resulted in a wrong tone
pitch.
This patch adds the calculation of the standard HD-audio type.
Also clean-up the fields in hda_beep struct.
Reference: bko#13162
http://bugzilla.kernel.org/show_bug.cgi?id=13162
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add proper prefix to each kernel message in hda_intel.c.
Also, avoid the unneeded prefix when CONFIG_SND_VERBOSE_PRINTK is used
together with snd_print*().
Reference: bko#13207
http://bugzilla.kernel.org/show_bug.cgi?id=13207
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support for MacBook 3,1 sound by adding a model new
"mb31" with the appropriate init verbs, mixers and channel modes to
the ALC883 configuration. patch_alc882() and patch_alc883() are
modified to handle the MacBook 3,1 sound-chip (Realtek ALC889A)
correctly.
Signed-off-by: Torben Schulz <public@letorbi.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCI entries of Creative with HD-audio class can be the devices
with emu20k1/emu20k2 chips. These are supported better by snd-ctxfi
driver. With that driver, the device will mutate from HD-audio to
its native class.
This patch adds a simple ifdef to avoid the conflict of device probe
between snd-hda-intel and snd-ctxfi drivers. 1102:0009 seems still
OK to be added as it has no emu20kx chip, and is a pure HD-audio
device.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Split the name string in hda_codec struct to vendor_name and chip_name
strings to be stored directly from the preset name.
Since mostly only the chip name is referred in many patch_*.c, this
results in the reduction of many codes in the end.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a mixer control for the STAC92XX boards to control the
DC bias of mic ports, allowing recording from both powered and
non-powered sources. It replaces the "Mic Output Switch" with "Mic Jack
Mode" to switch between Mic, Line In, and Line Out.
Signed-off-by: Nickolas Lloyd <ultrageek.lloyd@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the codec SSID for MacBook Pro 5,1 as compatible as MP51.
However, the headphone auto-muting function doesn't work. So,
this is just a tentative solution.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The model=acer for ALC883/889 doesn't work well for the recent Acer
Aspire laptops. Since model=auto works better nowadays, it's safer
to use the default fallback instead of the Acer specific one.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add specific configuration for Samsung NC10 mini notebook. Internal
mic/speakers will be correctly muted when plugging in external ones.
Mixer controls are added for speakers, headphones and PC beep.
"Boost" is added for the microphones.
Signed-off-by: Chris Pockelé <chris.pockele.f1@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the missing descriptions and the model names for Realtek codecs
to the documentation and the config table.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Most of unsol handlers defined in patch_realtek.c can be classified to
two types, mute via amp of pins and mute via ctl bits of pins.
Thus there are a big room to generalize each implementation.
This patch creates two generic functions, alc_automute_amp() and
alc_automute_pin(). The latter is actually changed from the previous
alc_sku_automute(). Each caller needs to initialize hp_pins and
speaker_pins properly at own init_hook.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The speaker auto-muting per HP plugging for ALC262 HIPPO and compatible
devices is slightly buggy as the "Master" or "Front" mixer control can
still toggle the speaker output even if the headphone is plugged.
This patch fixes the issue, and clean up the hippo-related codes
together with fixes of some inconsistent mixer names.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
VIA VT1708S and VT1702 codecs can have two SPDIF outputs. One of them
should have been handled as the extra digital out, but it's not
properly accessed.
This patch fixes the handling of the secondary SPDIF on these codecs
with the slave dig-out as found in patch_sigmatel.c. This makes the
use of such a device easier (for normal users).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BIOS on Mac Mini Core2 Duo sets both INPUT and OUTPUT pinctl bits to
the line-in jack, and it confuses the driver as if it's a valid input.
This patch adds the check of OUTPUT bit so that the driver fixes the
invalid pin setup.
Tested-by: Tino Keitel <tino.keitel@gmx.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commmit dfed0ef9b3 was reverted
accidentally by the merge of auto-detection fix patch.
Fixed again now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Enable auto-muting in model=auto only for devices with HP and speakers.
For devices with HP and line-outs, don't enable the auto-muting.
Also, add a debug print to show the auto-mute feature.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
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>
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>
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>
* '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>
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>
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>
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>
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>
