why pulseaudio schedule a delay of 28.784 seconds for playback?
Apr 3 09:29:45 quadrio pulseaudio[2036]: [alsa-sink-CX20561 Analog]
alsa-sink.c: Scheduling delay of 28784.48ms > 14.11ms, you might want to
investigate this to improve latency...
Apr 3 09:29:45 quadrio pulseaudio[2036]: [alsa-s
https://wiki.ubuntu.com/Audio/AlsaInfo
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Title:
volume of the center channel and subwoofer always 100% on creative
x-fi 5.1
To manage notificat
[ 15.900031] hda-intel :04:01.0: Codec #0 probe error; disabling
it...
[ 20.936039] hda-intel :04:01.0: no codecs initialized
** Summary changed:
- PCI/internal sound card not detected
+ PCI/internal sound card not detected[ 20.936039] hda-intel :04:01.0: no
codecs initialized
** Changed in: alsa-driver (Ubuntu)
Status: New => Invalid
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Title:
Home Care Program - Sensitive Skin
To manage notifications about this
Try specify model=6stack-dig
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/Documentation/sound/alsa/HD-Audio-Models.txt
00:1b.0 0403: 8086:2668 (rev 04)
Subsystem: 147b:1045
send email to alsa-devel mailing list to add pci quirk since Bios does
not setup pin default
https://bugs.launchpad.net/ubuntu/+source/alsa-
driver/+bug/1252733/comments/24
does it help if you disable the audio mixer ?
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Title:
[Dell Prec
https://bugs.launchpad.net/ubuntu/+source/alsa-
lib/+bug/1286021/comments/2
pulseauduo verbose log
https://wiki.ubuntu.com/PulseAudio/Log
how did you get it working on 14.04 ?
you have to post output of alsa-info.sh when sound is working and not
working, use diff to find out the difference of t
post pulseaudio verbose log
https://wiki.ubuntu.com/PulseAudio/Log
headphone and line out share the front playback volume control
control.31 {
iface CARD
name 'Line Out Jack'
value false
comment {
access r
https://bugzilla.kernel.org/show_bug.cgi?id=75151
** Bug watch added: Linux Kernel Bug Tracker #75151
http://bugzilla.kernel.org/show_bug.cgi?id=75151
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sound preference change the default sink and source
the problem is why iec958 was selected by pulseaudio if it has lower
priority than the other
do sound preference keep your setting after reboot ?
active profile:
sinks:
alsa_output.pci-_00_1f.5.iec958-ster
http://www.asus.com/Notebooks_Ultrabooks/N550JV/specifications/
1x combo audio jack
is the icon near the jack look like a headset or
headphone / Mic icons ?
you need line in for recording from piano 's line out
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https://wiki.ubuntu.com/PulseAudio/Log
post pulseaudio verbose log while you change volume from Max to 20%
do you mean emu10k1 's min dB -9.99 is lower than pulseaudio 's min
dB ?
state.Audigy {
control.1 {
iface MIXER
name 'PCM Front Playback Volume'
** Changed in: pulseaudio (Ubuntu)
Status: New => Incomplete
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Title:
Can't install linux-restricted-modules
To manage notifications about
** Changed in: pulseaudio (Ubuntu)
Status: New => Incomplete
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Title:
No change to volume in lowest 20% of control
To manage notifications
it is unlikely front Mic is selected automatically
there are four capture source
control.21 {
iface MIXER
name 'Capture Source'
value Line
comment {
access 'read write'
type ENUMERATE
Simple mixer control 'Channel Mode',0
Capabilities: enum
Items: '2ch' '4ch' '6ch'
Item0: '6ch'
when rear Mic and line in are retasked
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda/patch_via.c?id=f1e762ddc8035f803d15f8c524151bf788482ec2
there are five
you have to provide log to find out how pulseaudio calculate 20%
some of your volume controls are not defined in
http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mixer/paths
Simple mixer control 'PCM Front',0
Capabilities: pvolume
Playback channels: Front Left - Front
but your pulseaudio is 100% 0dB
index: 3
name:
driver:
flags: HARDWARE HW_VOLUME_CTRL LATENCY DYNAMIC_LATENCY
state: SUSPENDED
suspend cause: IDLE
priority: 9059
volume: 0: 100% 1: 100%
balance 0.00
base volume: 1
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/pci/intel8x0.c
static struct snd_pci_quirk intel8x0_clock_list[] = { SND_PCI_QUIRK(0x0e11,
0x008a, "AD1885", 41000), SND_PCI_QUIRK(0x1014, 0x0581, "AD1981B", 48000),
SND_PCI_QUIRK(0x1028, 0x00be, "AD1885", 44100), SND_PCI_Q
!!AC97 Codec information !!--- --startcollapse-- 0-0/0:
Analog Devices AD1980
PCI Subsys Vendor: 0x1028
PCI Subsys Device: 0x0177
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https://www.virtualbox.org/report/27?sort=modified&asc=0&page=1
It is because virtualbox change enumerated ac97 from stac9200 in
virtualbox 4.x to ad1980 in virtualbox 5.x
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https://www.virtualbox.org/browser/vbox/trunk/src/VBox/Devices/Audio/DevIchAc97.cpp
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Title:
[ICH - Intel 82801AA-ICH, playback] No sound at all
https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1487168
The emulated ac97 codec is different from the real ac97 codecs on dell
dimension 8400
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https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1065812
The real ad1980 support 5.1
Extended ID : codec=0 rev=0 AMAP LDAC SDAC CDAC DSA=0 DRA VRA
Extended status : LDAC SDAC CDAC VRA
PCM front DAC : 48000Hz PCM
Surr DAC : 48000Hz
PCM LFE DAC : 48000Hz
PCM ADC : 44100Hz
but enumer
AD18XX configuration Unchained : 0x1000,0x,0x Chained :
0x,0x,0x
The enumerated codec response incorrectly and appear as three codecs
AD18XX configuration Unchained : 0x1000,0x4000,0x2000 Chained :
0x,0x,0x
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https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/intel8x0.c?id=2b3b5485aa96d18b0025dfb2bc92c824dc81a780
ALSA: intel8x0: implement ac97_clock whitelist
The AC97 clock detection is not accurate in some cases. This patch adds
an initial whitelist for audio devices gather
[ 6.559582] snd_intel8x0 :00:05.0: white list rate for 1028:0177 is
48000
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Title:
[ICH - Intel 82801AA-ICH, playback] No sound at all
To man
** Also affects: alsa-driver via
http://www.virtualbox.org/ticket/15673
Importance: Unknown
Status: Unknown
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Title:
[ICH - Intel 828
https://www.virtualbox.org/log/vbox/trunk/src/VBox/Devices/Audio/DevIchAc97.cpp?rev=62355
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Title:
[ICH - Intel 82801AA-ICH, playback] No sound at
static struct snd_pcm_hardware snd_intel8x0_stream = { .info =
(SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), .formats =
SNDRV_PCM_FMTBIT_S16_LE,
.rates = SNDRV_PCM_RATE_48000,
.rat
udio] alsa-mixer.c: Volume element Master has 8 channels. That's too
much! I can't handle that!
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Title:
Xonar DSX AV200 not adjustable sound leve
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/refs/tags
You need to know ubuntu 16.04 using which version of kernel
You will find those XRUN/BUGS/invalid position message in system log
before patching and no more message after patching
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FIG. 15e is a diagrammatic block diagram of a control register of FIG.
15d, in accordance with one embodiment. Control register 384(c) includes
a sub-buffer enable interrupt flag 384(c)(0), a stop flag 384(c)(1) and
a next segment pointer 384(c)(2). Sub-buffer enable interrupt flag
384(c)(0) typica
You can specify card to just build the needed driver or specify all
cards
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Title:
snd_au8820 :01:04.0: BUG: , pos = 16386, buffer size = 1638
For building alsa driver as module
https://bugs.launchpad.net/ubuntu/+source/alsa-
driver/+bug/1551041/comments/71
You just follow the instruction in that link
What you need to do is
cd alsa-driver
git checkout build
utils/setup-alsa-kernel -c. ../../sound
./gitcompile --enable-dynamic-minors
A real kernel oops usually consist of stacktrace and dump of all cpu
registers
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Title:
snd_au8820 :01:04.0: BUG: , pos = 16386, buffer size =
There is no pcm devices for your card 2 usb midi
Pulseaudio should skip the card Similar to thinkpad ec coonsole at card
29 instead of reporting fail to find working profile
crw-rw+ 1 root audio 116, 2 Mar 18 08:04 /dev/snd/controlC0 crw-
rw+ 1 root audio 116, 12 Mar 18 08:04 /dev/snd/con
https://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mixer
/profile-sets/90-pulseaudio.rules
Just add your usb ssid and ENV{PULSE_IGNORE}="1" after those usb devices
You need udev daemom reload the rules or reboot
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In theory , there is no need to use any patch in oss directory if you
have disabled oss enumeration by passing --with-oss=no
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/log/sound/core/oss
You get checkout or use browser to extract the old version of two file
pcm_oss.c and mixer_
utils/setup-alsa-kernel -c ../../sound
Only use the following option to build debug driver which enable those dev_dbg
call
--with-debug=full
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[ 9.133753] snd_hda_intel :00:1f.3: azx_get_response timeout,
switching to polling mode: last cmd=0x000f
[ 10.142506] snd_hda_intel :00:1f.3: No response from codec, disabling
MSI: last cmd=0x000f
snd_hda_intel :00:1f.3: Codec #0 probe error; disabling it...
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Usually the reason fail to compile are those patch in acore directory
are out dated and you need to report to alsa-devel mailng list
https://git.kernel.org/cgit/linux/kernel/git/tiwai/alsa-driver-
build.git/tree/alsa/acore?h=build
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https://git.kernel.org/cgit/linux/kernel/git/tiwai/alsa-driver-
build.git/log/alsa/gitcompile?h=build
gitcompile only pass the command to configure and run make if no errors
./configure --help
You have to ask the author
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As Three input jacks of your onboard alc887 were unplugged and
pulseaudio does not regarded CD as available, pulseaudio may switch to
cmi8378 which has no capture channel map
control.40 { iface CARD name 'Rear Mic Jack' value false comment {
access read type BOOLEAN count 1 } } control.41 { iface
https://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules
/module-udev-detect.c
Can you post output of alsa-info.sh for your usb midi?
udev-detect /switch-on-connect need to skip any usb card without pcm
device?
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Did you change to alsa directory after git checkout build?
The alsa directory should contain
https://git.kernel.org/cgit/linux/kernel/git/tiwai/alsa-driver-build.git/tree/alsa?h=build
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https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound?id=f49921b894ec338696e47d985acc09df3cddbfd1
Seem no channel map for playback and capture device for your cmi8738, the
playback channel map is device 2
state.CMI8738 {
control.1 { iface PCM device 2 name 'Playback Channel
http://git.alsa-project.org/?p=alsa-
lib.git;a=blob;f=src/conf/cards/CMI8738-MC6.conf;hb=HEAD
Seem use different device for front, surround40 and surround51
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/pci/cmipci.c
You need to add channel map in snd_cmipci_pcm_new, sn
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound?id=2d3391ec0ecca37efb6bc995906292f47522b471
The difference between std channel map and alt channel map
You need to modify the driver by adding lines without "+" at the end of the two
routines
+ err = snd_pcm_add_ch
Can you post the pulseaudio verbose log?
You have to ask pulseaudio developer whether pulseaudio support spdif
input since your onboard have digital capture device, iec958 capture
switch and capture channel map for device 1
ARECORD List of CAPTURE Hardware Devices
card 1: MID [HDA In
https://wiki.ubuntu.com/PulseAudio/Log
https://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mixer/paths
Only iec958-stereo-output.conf but no iec958-stereo-input.conf
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https://git.kernel.org/cgit/linux/kernel/git/tiwai/alsa-driver-
build.git/log/?h=build
Last update was 2015 october 12
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Title:
snd_au8820 :0
https://git.kernel.org/cgit/linux/kernel/git/tiwai/alsa-driver-
build.git/tree/alsa/gitcompile?h=build
The minimum version of the packages are defined at the first few lines
automake
autoconf
libtool
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!!---Mixer controls for card 1 [U610]
Card hw:1 'U610'/'BEHRINGER UMX 610 at usb-:00:12.0-1, full speed'
Mixer name : 'USB Mixer'
Components : 'USB1397:01b0'
Controls : 0
Simple ctrls : 0
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You have to send email with the errors to Tiwai
The errors related to file in alsa-driver/alsa/acore are related to
changes in sound/core, you need his help to regenerate those patch files
in acore
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/log/sound/core
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gitcompile.sh and utils subdirectory
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Title:
snd_au8820 :01:04.0: BUG: , pos = 16386, buffer size = 16384,
period size = 1024
To manage no
The above link is just the oss directory just before
ALSA: oss: underflow in snd_mixer_oss_proc_write(). 2015 october 15
Save the plain text of the two affected files
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sys/class/sound/hwC0D2/init_pin_configs:
0x11 0x40f0
0x12 0x41f0
0x14 0x01014410
0x15 0x01011412
0x16 0x01016411
0x17 0x01012414
0x18 0x01a19c40
0x19 0x02a19c50
0x1a 0x0181344f
0x1b 0x02214c20
0x1c 0x593301f0
0x1d 0x4005e601
0x1e 0x014b6130
0x1f 0x01cb7160
/sys/class/sound/hw
https://www.freedesktop.org/software/systemd/man/udevadm.html
Just add
LABEL="pulseaudio_check_usb"
+ATTRS{idVendor}=="1397", ATTRS{idProduct}=="01b0",
ENV{PULSE_IGNORE}="1"
ATTRS{idVendor}=="17cc", ATTRS{idProduct}=="1978", ENV{PULSE_PROFILE_SET
}="native-instruments-audio8dj.conf"
udevadm c
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/core/oss?id=ab30965d9bfcd04931f9b70d00faa2ea614835a8
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Title:
snd_au8820 0
it is strange that pcm_avail.c which only wite one buffer can trigger
when the messages are rate limited
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/core/pcm_lib.c?id=74d779ab7c9f9024cfead259206e0e0b20ee37e4
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https://git.kernel.org/cgit/linux/kernel/git/tiwai/alsa-driver-
build.git/tree/alsa/gitcompile?h=build#n59
It need aclocal and autoconf to generate configure script
Have you installed the packages?
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It is unlikely that your patch fix the problem since none of the drivers
of those users have a volume control with multi channel
Only two users have drivers which does not have capture channel map (e.g.
cmipci and intel8x0)
it is not clear why hda-intel can trigger the bug, the acer aspire 7540
You need to open bug report at bugzilla.kernel.org as your snd-cmipci
driver does not have dB ranges and channel map, support 5.1 multi
channels but no volume control for rear center and lfe
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It seem there is no way to build alsa driver as module after kenerl 4.3
since alsa-driver-build no longer maintained after 2015 october
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Title:
s
Mar 21 12:40:41 hostname kernel: usb 3-2: 1:1: usb_set_interface failed (-110)
Mar 21 12:40:41 hostname kernel: usb 3-2: 1:1: usb_set_interface failed (-71)
Mar 21 12:40:41 hostname kernel: usb 3-2: 1:1: usb_set_interface failed (-19)
Seem no volume control
Card hw:4 'VX2'/'TOPPING VX2 at usb-
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/pci/cmipci.c
CMIPCI_SB_INPUT_SW("Line Capture Route", 4, 3),
You have to ask the Maintainter of cmipci why the control have 4 values if it
emulate sb16
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http://www.cmedia.com.tw/productsdetail/page-p/c1serno-25/c2serno-28/pserno-31.html
8738-LX/MX supports 6CH DAC 5.1 surround output for movies and 3D games.
8738-SX supports 4CH DAC surround output for movies and 3D games.
Built-in 2CH 16bit ADC.
Supports full-duplex 44.1K/48KHz sample rate playba
https://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/src/modules/alsa
/alsa-mixer.c?id=c87917cf0da9add5c282a1c15831fd481f9fcc22
Pulseaudio drop support of those volume controls with more than two
channels
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-32 is EPIPE
http://git.alsa-project.org/?p=alsa-lib.git;a=blob;f=src/pcm/pcm.c
http://git.alsa-project.org/?p=alsa-lib.git;a=blob;f=src/pcm/pcm_hw.c
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Try specify period time to multiple of 10ms since you are using 44100Hz
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Title:
OpenAL Software Silent/Freezes
To manage notifications about thi
Your usb audio are using about 25ms period time with 4 periods
) I: [pulseaudio] source.c: device.vendor.id = "0d8c" ( 5.364| 0.000) I:
[pulseaudio] source.c: device.vendor.name = "C-Media Electronics, Inc."
( 5.364| 0.000) I: [pulseaudio] source.c: device.product.id = "0102" (
5.364| 0.000) I: [p
Blender request 15ms but your usb use period time 24.998 ms
0) I: [pulseaudio] sink-input.c: application.process.binary = "blender" (
6.034| 0.000) I: [pulseaudio] sink-input.c: application.language = "C" ( 6.034|
0.000) I: [pulseaudio] sink-input.c: window.x11.display = ":0.0" ( 6.034|
0.000)
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/log/sound/pci/hda/thinkpad_helper.c
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/log/sound/pci/hda/patch_realtek.c?qt=grep&q=alc269
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https://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/daemon/daemon.conf.in
; default-fragments = 4
; default-fragment-size-msec = 25
https://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/?id=810aa36189f82f77403826defa76a4f9f1f01454
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** Bug watch added: Linux Kernel Bug Tracker #115531
http://bugzilla.kernel.org/show_bug.cgi?id=115531
** Also affects: alsa-driver via
http://bugzilla.kernel.org/show_bug.cgi?id=115531
Importance: Unknown
Status: Unknown
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Which values do you use to fix the problem?
Blender require 60ms but your usb are configured at 100ms
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Title:
OpenAL Software Silent/Freezes
To
You need to remove reduant mic/line in by hdajackretask
[ 9.704181] snd_hda_codec_realtek hdaudioC1D0: autoconfig for ALC282:
line_outs=1 (0x1b/0x0/0x0/0x0/0x0) type:line
[ 9.704186] snd_hda_codec_realtek hdaudioC1D0: speaker_outs=1
(0x14/0x0/0x0/0x0/0x0)
[ 9.704190] snd_hda_codec_realtek hda
control.14 { iface MIXER name 'Capture Source' value Mic comment {
access 'read write' type ENUMERATED count 1 item.0 'Internal Mic' item.1
Mic item.2 'Mic 1' item.3 Line } }
You need to select mic 1
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control.24 { iface CARD name 'Mic Jack' value false comment { access
read type BOOLEAN count 1 } } control.25 { iface CARD name 'Mic Jack'
index 1 value true comment { access read type BOOLEAN count 1 } }
control.26 { iface CARD name 'Line Jack' value false comment { access
read type BOOLEAN count
** Changed in: alsa-driver (Ubuntu)
Status: New => Incomplete
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Title:
[TMP645-M, Realtek ALC282, Pink Mic, Left] No sound at all
To manag
You have to file upstream bug report at bugzilla.kernel.org since driver
one hp jack to line out, one of them may be dock hp
0x1b [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out
Control: name="Line Out Playback Switch", index=0, device=0
ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps:
You must ask the author and reporter since the patch change the pin
default as your vgn fs315 have same pci ssid 0x81bb as his sony laptop
** Changed in: alsa-driver (Ubuntu)
Status: New => Incomplete
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Bugs, which is
The point is laptop should have speaker pin, line out at sep mean dock
line out
Pulseaudio rely on the jack state of headphone to select active port and
mute/unmute
You have to provide out put of
hdajacksensetest -a
to find out which node is headphone
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https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda?id=c3c2c9e7ff3e38bd9ff5b721b6ae8634fce42802
it worked in the previous basic model because the driver enable output
on four pins,
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* select line-out */
Seem only blue line in jack was plugged and green line out jack was not
plugged
analog-input-linein: En línea (priority 8100, latency offset 0 usec, available:
yes) properties:
analog-output-lineout: Línea de salida (priority 9900, latency offset 0
usec, available: no) properties:Se
** Chang
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/log/sound/pci/hda/patch_cirrus.c?qt=grep&q=mic
** Changed in: alsa-driver (Ubuntu)
Status: New => Incomplete
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https://b
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda/patch_cirrus.c?id=ffe4d12b151018e831f0ca2e05fe78a55238c517
Try the step in the bug report to verify that you need similar fix
: https://bugzilla.kernel.org/show_bug.cgi?id=50781
** Bug watch added: Linux Kernel B
Which node change to No when you unplugged the headphone and mic jack
https://sourceforge.net/p/alsa/mailman/message/31661966/
Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch
Playback channels: Front Left - Front Right Limits: Playback 0 - 64
Mono: Front Left: Playback 0 [0%] [
Without specify any model, try record stereo using audacity using alsa
hw device to find out whether you get both left and right, just left or
just right channel from the display wave signal
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You have to post output of alsa-info.sh and pactl list
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https://bugs.launchpad.net/bugs/1566596
Title:
MBP 8,2 internal microphone stopped working
To manage notifications about this bu
You have line in jack and internal mic, but power setting are D3 instead
of D0
You need to file upstream bug report
It is strange that mic phantom jack is created instead of internal mic
phantom jack
control.22 { iface CARD name 'Mic Phantom Jack' value true comment {
access read type BOOLEAN co
Can you record stereo from line in jack?
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https://bugs.launchpad.net/bugs/1566596
Title:
MBP 8,2 internal microphone stopped working
To manage notifications about this bug go to:
https
https://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mixer/paths
/analog-input-internal-mic.conf
Pulseaudio expect internal mic phantom jack and seem missing
jack line
state.unplugged=unknown
https://bugzilla.kernel.org/enter_bug.cgi?product=Drivers
Sound (alsa)
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You
- line_in_auto_switch (bool): enable/disable the line-in auto-switch
feature; default false
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/Documentation/sound/alsa
/HD-Audio.txt
You need hint to enable line in auto switch when there is no mic jack
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It is bug of driver since it should create capture source which allow
you to select line in and internal mic when line in auto switch is
disabled by default
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** Also affects: alsa-driver via
http://bugzilla.kernel.org/show_bug.cgi?id=116051
Importance: Unknown
Status: Unknown
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https://bugs.launchpad.net/bugs/1566596
Title:
MBP 8
4.697026] intel_sst_acpi 808622A8:00: No matching machine driver found
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/soc/intel/common/sst-acpi.c#n102
mach = sst_acpi_find_machine(desc->machines);
if (mach == NULL) {
dev_err(dev, "No matching ASoC machine driver found
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/soc/intel/common
/sst-match-acpi.c?id=1fdb7c14a24689e9c7ec51aa16a3970a7ec0fdde
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https://bugs.launchpad.net/bugs/15
http://git.kernel.org/cgit/linux/kernel/git/tiwai/hda-
emu.git/plain/codecs/cs4206-macbook-pro-81?id=HEAD
it is possiblw that analog output has 13 samples delay but digital
output has 4 sample delay for those macbook using cs4206
ode 0x02 [Audio Output] wcaps 0xd041d: Stereo Amp-Out
Control: n
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