http://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda/hda_jack.c?id=a9c74173f4be2a536ef8d8c88642e41527f2d8b4
you need to ask the author about the definition of is_jack_detectable
function
the pincap of HP and mic pin of your vt1708 support
AC_PINCAP_PRES_DETECT but
you have to run the script and post the output
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Title:
[CF-74ECBAXBM, SigmaTel STAC9200, Speaker, Internal] Playback problem
To manage
the volume control seem have large gain (0x2e - 0x11)
you have to post the pulseaudio verbose log
did the crackling occur when capture volume is over 0dB ?
post the output of alsa-info.sh
Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In
Control: name=Capture Switch, index=0, device=0
Node 0x10 [Audio Output] wcaps 0x411: Stereo
Device: name=VT1708 Analog, type=Audio, device=0
Converter: stream=8, channel=0
PCM:
rates [0x7e0]: 44100 48000 88200 96000 176400 192000
bits [0xa]: 16 24
formats [0x1]: PCM
does the driver still support 24 or 32 bit ?
speaker-test
need to specify -D hw:0
speaker-test -c2 -Dhw:0 -t sine -F S16_LE
speaker-test -c2 -Dhw:0 -t sine -F S24_LE
speaker-test -c2 -Dhw:0 -t sine -F S32_LE
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[ 9548.424755] hda_codec: invalid CONNECT_LIST verb 12[2]:2100
Node 0x12 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out
Control: name=Capture Volume, index=0, device=0
ControlAmp: chs=3, dir=Out, idx=0, ofs=0
Control: name=Capture Switch, index=0, device=0
ControlAmp: chs=3, dir=Out,
[45722.684940] hda_codec: invalid CONNECT_LIST verb 12[2]:2100
[45812.811552] hda_codec: invalid CONNECT_LIST verb 12[2]:2100
Node 0x12 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out
Control: name=Capture Volume, index=0, device=0
ControlAmp: chs=3, dir=Out, idx=0, ofs=0
Control:
is there any specific reason for you to disable the on board sound card
?
index=-2 is a specific option to force the card to not loaded as card 0
e.g. intel8x0-modem have to loaded after snd-intel8x0
[42181.620791] usb 1-1.1.2: Product: USB Audio DAC
[42181.620792] usb 1-1.1.2:
It is strange that Mic boost control is at node 0x2b since wcaps does
not support Amp-in
however dB range of capture volume control is -16.5 to 30 dB
Node 0x10 [Audio Input] wcaps 0x10051b: Stereo Amp-In
Control: name=Capture Volume, index=0, device=0
ControlAmp: chs=3, dir=In, idx=0,
do you mean that you want to limit the dBmax to 0 instead of 30 ?
Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1
Amp-In vals: [0x1e 0x1e]
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seem strange to have PCM Loopback Playback Volume and Mic Playback
Volume at same index but 0x2a, 0x2b and 0x29 are input pin complexes
Node 0x21 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In
Control: name=PCM Loopback Playback Volume, index=0, device=0
ControlAmp: chs=3, dir=In, idx=3,
Node 0x24 [Pin Complex] wcaps 0x40050d: Stereo Amp-Out
Control: name=Speaker Playback Switch, index=0, device=0
ControlAmp: chs=3, dir=Out, idx=0, ofs=0
Control: name=Speaker Phantom Jack, index=0, device=0
Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-Out vals: [0x00
you have to ask the author why the driver create the headphone jack
detection control if the driver should *NOT* do detection
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=d39a3ae8215ad90e68ca5fbbced507b07052018b
+ case HDA_FIXUP_ACT_PROBE:
+ /* CF-74 has no headphone
you have to assign index=0 to usb audio and index=1 to your onboard
sound card if you want to your usb audio assign to card 0
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Title:
USB audio
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/log/sound/pci/hda/patch_sigmatel.c?qt=grepq=panasonic
it is strange that there is driver pin Fishpond but the model Panasonic seem
didiot not has any pin fixup
Files
!!---
/sys/class/sound/hwC0D0/init_pin_configs:
0x08
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda/patch_sigmatel.c?id=49c605db8d75216f88c3a57cfed3adfaddb71c6a
seem you are not using latest alsa driver
it used ref9200_pin_config fixup but there is no v.pins
control.14 {
iface CARD
name 'Speaker Front Jack'
value false
comment {
access read
type BOOLEAN
count 1
}
}
control.15 {
The name should be Internal Mic Boost Capture Volume instead of
Internal Mic Boost Volume since node 0x30 is only connected to [Audio
Input] and there is no audio path from node 0x30 to output pins
capabilities should be cvolume instead of volume
Simple mixer control 'Internal Mic Boost',0
how did you perform the recording test ? ( through pulseaudio or using
hw:0 )
you have to provide pulseaudio verbose log if you perform record through
pulseaudio
you have to provide output of alsa-info.sh or amixer instead of screen
shot as your screen shot does not provide dB value of the
to verify that there node 0x3e is a hidden widget
use hda-verb to get the widget cap and check the return value to find
out node 0x3e is [audio mixer]/[audio selector] or not exist
send: NID=0x3e, VERB=0xf00(get_parameters), PARM=0x9(audio_wid_cap)
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ftp://download.nvidia.com/XFree86/gpu-hdmi-audio-document/gpu-hdmi-audio.html
but ad1989b is your on board sound
Node 0x1b [Pin Complex] wcaps 0x40030d: Stereo Digital Amp-Out
Control: name=IEC958 Playback Volume, index=0, device=0
ControlAmp: chs=3, dir=Out, idx=0, ofs=0
Amp-Out caps:
you are using stereo analog output
Default sample spec: s16le 2ch 44100Hz
Default channel map: front-left,front-right
Default sink name: alsa_output.pci-_00_1b.0.analog-stereo
Default source name: alsa_input.pci-_00_1b.0.analog-stereo
try latest alsa driver
do your headphone get any sound ?
http://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda/patch_sigmatel.c?id=2acc9dcb609427a20463e529ef552dd93b469847
ALSA: hda - Fix silent headphone output on Panasonic CF-74
CF-74 does the headphone/speaker switching on hardware, thus
did this HDMI playback volume affect the volume ?
Simple mixer control 'HDMI',0
Capabilities: pvolume penum
Playback channels: Front Left - Front Right
Limits: Playback 0 - 39
Mono:
Front Left: Playback 39 [100%] [0.00dB]
Front Right: Playback 39 [100%] [0.00dB]
7.3.4.6 Audio Widget Capabilities The Audio Capabilities control returns
a set of bit fields describing the audio capabilities of the widget.
Parameter ID: 09h
Response Format:
23:20 Type
10 Power Cntrl
8 Conn List
0 Channel count
Figure 86. Audio Widget Capabilities Response Format
send: NID=0x3e, VERB=0xf00(get_parameters), PARM=0xe(connect_len)
send: NID=0x3e, VERB=0xf02(get_connect_list), PARM=0x0
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Title:
System76
is this widget similar to
http://git.kernel.org/cgit/linux/kernel/git/tiwai/hda-
emu.git/plain/codecs/vt1812-clevo-b5130m?id=HEAD
Node 0x3e [Audio Mixer] wcaps 0x200500: Mono
Power states: D0 D1 D2 D3
Power: setting=D0, actual=D0
Connection: 1
0x1c
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you/pulseaudio have selected digital output
Default sample spec: s16le 2ch 44100Hz
Default channel map: front-left,front-right
Default sink name: alsa_output.pci-_00_1b.0.iec958-stereo
Default source name: alsa_input.pci-_00_1b.0.analog-stereo
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the pulseaudio verbose show you how PA make up
Ports:
analog-output: Analog Output (priority: 9900)
Active Port: analog-output
change the volume control from the selected the output.conf
http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mixer/paths
when internal mic (0x23) is selected, the amp -in value from node 0x17
(mux) must be muted (similar to the value [0x80 0x80] from node 0x18 )
Node 0x14 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L
Control: name=Capture Volume, index=0, device=0
ControlAmp: chs=3, dir=In, idx=0, ofs=0
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda/patch_conexant.c?id=e4db0952e542090c605fd41d31d761f1b4624f4a
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echo 1 /sys/class/sound/hwC0D0/reconfig
the side effect of dynamic reconfigure is the driver remove the controls
(1 to 24) except those user space softvol controls (25 to 26), create
the new controls from 27
Alsactl output
!!--
--startcollapse--
state.PCH {
control.25 {
https://wiki.ubuntu.com/PulseAudio/Log
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Title:
[CF-74ECBAXBM, SigmaTel STAC9200, Speaker, Internal] Playback problem
To manage notifications
https://wiki.ubuntu.com/Audio/AlsaInfo
https://wiki.ubuntu.com/PulseAudio/Log
pactl list
** Changed in: pulseaudio (Ubuntu)
Status: Confirmed = Incomplete
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https://wiki.ubuntu.com/Audio/AlsaInfo
https://wiki.ubuntu.com/PulseAudio/Log
pactl list
** Changed in: pulseaudio (Ubuntu)
Status: New = Incomplete
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http://www.intel.com/support/motherboards/desktop/sb/CS-032871.htm
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https://bugs.launchpad.net/bugs/1227795
Title:
[P5Q DELUXE, Analog Devices AD1989B, Other SPDIF Out, Rear] No sound
why do you overwrite pcm.default and ctl.default if you are running
puseaudio server
pcm.!default {
type hw
card 1
}
ctl.!default {
type hw
card 1
}
the internal mic is not availbable when you plug the external mic ?
control.14 {
iface CARD
name
is there any reason to enable ucm ?
( 0.017| 0.000) D: [pulseaudio] module-udev-detect.c: Loading
module-alsa-card with arguments 'device_id=0 name=pci-_00_03.0
card_name=alsa_card.pci-_00_03.0 namereg_fail=false tsched=yes
fixed_latency_range=no ignore_dB=no deferred_volume=yes
the icon near the Jack indicate it is a headset jack (headphone with
Mic)
http://www.dell.com/us/dfh/p/xps-13-l321x-mlk/pd
are you using headphone (TRS) or headset (TRRS) connector
post result of hda-jack-sense-test when headset is plugged and unplugged
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ALSA configuration files
!!---
!!System wide config file (/etc/asound.conf)
pcm.!default {
type hw
card 1
}
ctl.!default {
type hw
card 1
}
pulseaudio] module-alsa-card.c: Jack 'Headphone Jack' is now unplugged
( 0.174| 0.000) D: [pulseaudio] module-alsa-card.c: Jack 'Speaker Phantom
Jack' is now plugged in
( 0.174| 0.000) D: [pulseaudio] module-alsa-card.c: Jack 'Headphone Jack'
is now unplugged
( 0.174| 0.000) D:
post the output of alsa-info.sh for working and non-working
diff -u working-info non-working-info
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Title:
[Fujitsu S7110,ALC 262] Not sound/
post the out of alsa-info.sh and pulseaudio verbose log
https://wiki.ubuntu.com/Audio/AlsaInfo
https://wiki.ubuntu.com/PulseAudio/Log
digital output was selected
Default sample spec: s16le 2ch 44100Hz
Default channel map: front-left,front-right
Default sink name:
may be the left and right of your internal Mic are in inverted phase
try alsamixer -c0 to mute the right channel of the capture switch
Simple mixer control 'Capture',0
Capabilities: cvolume cswitch
Capture channels: Front Left - Front Right
Limits: Capture 0 - 80
Front Left: Capture 56
https://launchpad.net/~diwic/+archive/hda
install snd-hday-tools
run and post the output of hda-jack-sense-test -a
when you plug and unplug the headset
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in patch_realtek.c
there is a snd_pci_quirk_vendor for all sony vaio to use
ALC269_FIXUP_SONY_VAIO
static const struct hda_fixup alc269_fixups[] = {
[ALC269_FIXUP_SONY_VAIO] = {
.type = HDA_FIXUP_PINCTLS,
.v.pins = (const struct hda_pintbl[]) {
ftp://download.nvidia.com/XFree86/gpu-hdmi-audio-document/gpu-hdmi-
audio.html#_verify_your_eld_is_valid
do you have valid ELD ?
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Title:
you need root privilege
sudo hda-Jack-sense-test -a -c1
Traceback (most recent call last):
File /usr/bin/hda-jack-sense-test, line 47, in module
codec = HDACodec(cardindex, codecindex)
File /usr/share/snd-hda-tools/hda_codec.py, line 1023, in __init__
self.fd =
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/log/sound/pci/hda?qt=grepq=headset+mic
!Advanced information - PCI Vendor/Device/Subsystem ID's
!!
00:1b.0 0403: 8086:1e20 (rev 04)
Subsystem: 1028:058b
seem your
static const struct hda_model_fixup alc269_fixup_models[] = {
{.id = ALC269_FIXUP_AMIC, .name = laptop-amic},
{.id = ALC269_FIXUP_DMIC, .name = laptop-dmic},
{.id = ALC269_FIXUP_STEREO_DMIC, .name = alc269-dmic},
{.id = ALC271_FIXUP_DMIC, .name = alc271-dmic},
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda/patch_realtek.c?id=73bdd597823e2231dc882577dbbaf8df92fe1775
there are two models and you need to find out the pin detect of
headphone or MicJack is used when headset is pluuged using hda-jack-
sense-test
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you have to provide pulseaudio verbose log and output of alsa-info,.sh
when headphone is plugged if you still have problem with latest alsa
driver for the same motherobard
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can you post the pulseaudio verbose log
ports:
analog-output-speaker: Speakers (priority 1, latency offset
0 usec, available: unknown)
properties:
device.icon_name = audio-speakers
seem there are vendor specific coefficient for the headset on those dell
machines
but your alc286 is not in the patch
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda/patch_realtek.c?id=73bdd597823e2231dc882577dbbaf8df92fe1775
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http://mailman.alsa-project.org/pipermail/alsa-
devel/2013-September/066430.html
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Title:
[ASUS P8Z68-V PRO, Realtek ALC892, Green Line Out,
it depend on whether your computer chassis have HDA front audio panel
and the front audio panel connector are connected according to your
motherboard specification.
use ac97 front panel or connect the wrong pin of hda front audio panel
may lead to jack detection problem even when your headphone
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/plain/Documentation/sound/alsa
/HD-Audio.txt
try enable trace of hda_unsol_event , normally one unsol_event for the
plugged/un plugged event instead of flooding events
Tracepoints
~~~
The hd-audio driver gives a few basic
your log only show headphone was plugged in and no mic was plugged
are you plugged the headset during the log ?
5.172| 0.000) I: [alsa-sink-HDMI 0] alsa-sink.c: Device suspended...
( 5.173| 0.000) D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is
accessible: yes
( 5.173|
https://wiki.ubuntu.com/PulseAudio/Log
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Title:
[CF-74ECBAXBM, SigmaTel STAC9200, Speaker, Internal] No sound at all
To manage notifications
try to find out the VAL return by get pin sense after those unsolicited
events
`hda:hda_send_cmd` traces each CORB write while `hda:hda_get_response`
traces the response from RIRB (only when read from the codec driver).
The generic parser supports the following hints:
- trigger_sense (bool):
can you upload pulseverbose.log ?
your last post did not contain the whole log or you stop the log too
fast
the log should contain
[pulseaudio] main.c: Got org.pulseaudio.Server!
[pulseaudio] main.c: Daemon startup complete.
and those headphone plug and unplug event
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With firmware 1.107 and kernel 3.10, the green speaker out and silver
headphone out work, but the front headphone port still does not work at
all. I don't have any equipment for testing digital out though.
http://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=a73d511c4867c5aa75a9ab50f7e73d5086c48cda
HP is expected at node 0x10
#define UNSOL_TAG_HP0x10
#define UNSOL_TAG_AMIC1 0x12
#define UNSOL_TAG_DSP 0x16
+static void ca0132_init_unsol(struct hda_codec *codec)
+{
+
https://www.google.com/intl/en/chrome/devices/chromebook-pixel/#pixel-
specs
The Chormebook has a Combo headphone/mic jack which is different from those
pcie sound cards
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https://wiki.ubuntu.com/Audio/AlsaInfo
https://wiki.ubuntu.com/PulseAudio/Log
pactl list
pactl stat
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https://bugs.launchpad.net/bugs/826565
Title:
[EMU10K1 - SB Live! 5.1
http://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda/patch_sigmatel.c?id=2acc9dcb609427a20463e529ef552dd93b469847
ALSA: hda - Fix silent headphone output on Panasonic CF-74
CF-74 does the headphone/speaker switching on hardware, thus the driver
shouldn't do any
http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/?id=0b421f0d93f691ccd95d3999a1a9045caee76671
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Title:
[ASRock A75M-ITX, Realtek ALC892,
device.profile.name = iec958-stereo
device.profile.description = Digital Stereo (IEC958)
device.description = Built-in Audio Digital Stereo (IEC958)
alsa.mixer_name = Realtek ALC892
alsa.components = HDA:10ec0892,18498892,00100302
check whether your graphic driver can obtain EDID from your Sharp TV
and need to know your graphic driver pass valid ELD to the audio driver
https://bugzilla.kernel.org/show_bug.cgi?id=61921
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state.Generic {
control.1 {
iface CARD
name 'HDMI/DP,pcm=3 Jack'
value false
comment {
access read
type BOOLEAN
count 1
}
}
this
https://bugzilla.kernel.org/show_bug.cgi?id=48831#c2
** Bug watch added: Linux Kernel Bug Tracker #48831
http://bugzilla.kernel.org/show_bug.cgi?id=48831
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*** This bug is a duplicate of bug 1185417 ***
https://bugs.launchpad.net/bugs/1185417
** This bug has been marked a duplicate of bug 1185417
PCI/internal sound card not detected
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you must specify model=6stack since BIOS did not set up pin defaults
do this motherboard has any spdif (optical/coaxial) ? 6stack-dig
send email to alsa devel mailing list to add quirk for this motheboard
http://www.alsa-project.org/mailman/listinfo/alsa-devel
Advanced information - PCI
you should not overwrite pcm.default if you are using pulseaudio
!!ALSA configuration files
!!
!!System wide config file (/etc/asound.conf)
pcm.!default {
type plug
slave {
pcm iec958
}
}
!!Aplay/Arecord output
!!
APLAY
List of
[ 235.820137] snd-usb-audio 3-1:1.0: no reset_resume for driver snd-usb-audio?
[ 235.820139] snd-usb-audio 3-1:1.1: no reset_resume for driver snd-usb-audio?
[ 235.820140] snd-usb-audio 3-1:1.2: no reset_resume for driver snd-usb-audio?
[ 235.963224] PM: resume of devices complete after
you have to ask the Co-author (realteak) of the patch or Sony support
whether your alc268 have vendor specific coefficient to enable the
headset Michael feature
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post the pulseaudio log of both version (working and non working) for
plug and unplug the headphone
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https://bugs.launchpad.net/bugs/1203052
Title:
[ICH4 - Intel 82801DB-ICH4,
https://wiki.ubuntu.com/PulseAudio/Log
pulseaudio verbose log when you change the capture volume from min to
Max by in sound preference
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Title:
open a terminal and use alsamixer -c0 to monitor the changes in the
capture volume and boost while you change the capture slider in sound
preference
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** Changed in: pulseaudio (Ubuntu)
Status: Incomplete = Confirmed
** Package changed: pulseaudio (Ubuntu) = alsa-driver (Ubuntu)
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Title:
** Summary changed:
- No sound output: Audio directed to Dummy
+ [ABIT AA8, ALC880] No sound output: Audio directed to Dummy
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Title:
[ABIT AA8,
post the system log from suspend to resume
pactl stat before suspend and after resume
pactl list before suspend and after resume
if possible upgrade to latest driver since your ad1988b still not support jack
detection
unsolicited event still not enabled (no automute)
Node 0x11 [Pin
post the pulseaudio verbose log from suspend to resume
https://wiki.ubuntu.com/PulseAudio/Log
** Changed in: pulseaudio (Ubuntu)
Status: New = Incomplete
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** Changed in: alsa-driver (Ubuntu)
Status: New = Incomplete
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Title:
[ICH4 - Intel 82801DB-ICH4, playback] No sound at all after upgrade
is there any reason to unmute Mic playback volume ?
Simple mixer control 'Mic',0
Capabilities: pvolume pvolume-joined cvolume cvolume-joined pswitch
pswitch-joined cswitch cswitch-joined cswitch-exclusive penum
Capture exclusive group: 0
Playback channels: Mono
Capture channels: Mono
just edit the firmware file generated by hda-jack-retask and add the
hint
add_jack_modes to add Mic Jack Mode (Mic and Line in Switching) and Line
Out Jack Mode for (Line Out and HP switching)
Early Patching
~~
When CONFIG_SND_HDA_PATCH_LOADER=y is set, you can pass a patch as a
seem Jack detection problem, receive many plug and unplug events within
same second
try disable auto mute
or
set misc = NO_PRESENCE of pin default of headphone using hda-Jack-retask
ct 20 13:31:05 julpierto pulseaudio[1678]: ( 1.595| 1.595) [pulseaudio]
module-alsa-card.c: Jack 'Front
http://cgit.freedesktop.org/pulseaudio/pulseaudio/plain/src/tests
/volume-test.c
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Title:
sound is distorted
To manage notifications about this
when the on board sound card has hardware volume control, pulseaudio
provide info of hardware and calculated software volume
8| 0.000) D: [alsa-source] alsa-source.c: Requested volume: 0: 1% 1: 1%
( 70.068| 0.000) D: [alsa-source] alsa-source.c:in dB: 0:
-128.01 dB 1:
http://git.alsa-project.org/?p=alsa-
plugins.git;a=blob_plain;f=doc/a52.txt;hb=HEAD
The card option specifies the card ID or number of the SPDIF.
The output PCM becomes iec958:{CARD=$CARD} with extra AESx
settings. When omitted, the default card is used.
- The slavepcm option specifies a
do nimbus has optical spdif?
http://www.cirrus7.com/produkte/cirrus7-nimbus/leistungsmerkmale
Audio 2x Stereo 3.5mm Klinke, 1x digital optical S/PDIF
http://www.intel.com/content/www/us/en/motherboards/desktop-motherboards/desktop-board-dq77kb.html
in theory , Michael Jack can also be
do your motherboard have two spdif out ?
Node 0x0e [Audio Output] wcaps 0x611: Stereo Digital
Control: name=IEC958 Playback Con Mask, index=0, device=0
Control: name=IEC958 Playback Pro Mask, index=0, device=0
Control: name=IEC958 Playback Default, index=0, device=0
Control: name=IEC958
seem missing stereo mix
Node 0x21 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In
Control: name=Front Mic Playback Volume, index=0, device=0
ControlAmp: chs=3, dir=In, idx=3, ofs=0
Control: name=Front Mic Playback Switch, index=0, device=0
ControlAmp: chs=3, dir=In, idx=3, ofs=0
Too bad Ubuntu does not support digital surround 5.1 on the s/pdif
output.
http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/?id=814c0371a493cf47df4e6e27253cfd5ec3cef783
+ pa_snprintf(device_name, len, %s,AES0=6, u-device_name);
http://git.alsa-project.org/?p=alsa-
try latest alsa driver
https://wiki.ubuntu.com/Audio/UpgradingAlsa/DKMS
post the output of alsa-info.sh
pactl stat
pactl list
and
pulseaudio verbose log
https://wiki.ubuntu.com/PulseAudio/Log
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Bugs, which is
!!Advanced information - PCI Vendor/Device/Subsystem ID's
!!
00:1b.0 0403: 8086:1e20 (rev 04)
Subsystem: 8086:2036
if the PCI SSID cannot idenify it is a product of Cirrus 7
the driver only regard this is a computer using Intel
https://wiki.ubuntu.com/PulseAudio/Log
lsusb -
** Changed in: alsa-driver (Ubuntu)
Status: New = Incomplete
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Title:
[USB-Audio -
default channel of a52 is 6 , so your config fail when pulseaudio use
iec958-ac3-surround-40
D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_channels(4) failed:
Invalid argument
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!!---Mixer controls for card 2 [U46DJ]
Card hw:2 'U46DJ'/'ESI U46DJ at usb-:00:14.0-3, full speed'
Mixer name: 'USB Mixer'
Components: 'USB0a92:00e1'
Controls : 9
Simple ctrls : 4
Simple mixer control 'U46DJ',0
Capabilities: pvolume cvolume pswitch cswitch
how do those alsa control map to those windows controls in the user
manual ?
refer to user manual,
there are two input volume control with dB scale
three stereo output volume controls, mix and master volume controls
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