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>
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>
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>
