** Changed in: pulseaudio (Ubuntu)
Status: Expired => Confirmed
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https://bugs.launchpad.net/bugs/1259070
Title:
N90SV, Realtek ALC663, Mic, Internal Underruns, dropouts or
[Expired for pulseaudio (Ubuntu) because there has been no activity for
60 days.]
** Changed in: pulseaudio (Ubuntu)
Status: Incomplete = Expired
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The middle headphone jack has node 26
the front headphone jack is labeled spdif/headphone and has node 25
But I don't really understand what this has to do with my recording problems...
btw, recording from the microphone jack is noisy as well...
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if record sound is OK when using alsa hw:0,0 or disable timer
scheduling
this mean bugs in pulseaudio timer scheduling in recording
you have to provide DEBUG_TIMING pulseaudio log
** Package changed: alsa-driver (Ubuntu) = pulseaudio (Ubuntu)
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you need someone to provide a debugged version of pulseaudio
#define DEBUG_TIMING 1
in
http://cgit.freedesktop.org/pulseaudio/pulseaudio/plain/src/modules/alsa
/alsa-source.c
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: [alsa-source-ALC663 Analog] alsa-util.c: Trying to disable ALSA period
wakeups, using timers only
D: [alsa-source-ALC663 Analog] alsa-util.c: Maximum hw buffer size is 371 ms
D: [alsa-source-ALC663 Analog] alsa-util.c: Set buffer size first (to 16384
samples), period size second (to 8192
Just did some more testing:
when looking at device properties while recording, I had some strange
differences in buffer size etc.
parecord (which records fine) has properties like this:
43550 μs (= buffer: 0 μs + source: 43550 μs)
skype has properties:
752 μs (= buffer: 0 μs + source: 752 μs)
https://bugs.freedesktop.org/enter_bug.cgi?product=PulseAudio
as pulseaudio disable period wakeup when using timer scheduling
you can only use the following to find out whether the alsa period
update properly if you are using tsched=0
http://www.alsa-project.org/main/index.php/XRUN_Debug
8
try hda-analyzer
check and uncheck the Enable checkbox inside Digital Convertor of
node 0x6 or node 0x10 to find out which node can turn on/off the red
light of the spdif
Node 0x06 [Audio Output] wcaps 0x611: Stereo Digital
Converter: stream=0, channel=0
Digital:
Digital category: 0x0
** Attachment added: alsa-info.txt.9K9ilM72WW
https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1259070/+attachment/3927972/+files/alsa-info.txt.9K9ilM72WW
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** Attachment added: mono.wav
https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1259070/+attachment/3927974/+files/mono.wav
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Title:
Tried to enable/disable 0x6 and 0x10 in hda-analyzer. No lights - but I
wouldn't know where to look and whether there are any...
Recording in audacity using hw:0,0 is o.k. using pulse is awful.
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do you know which node is the headphone at the middle ?
amixer -c0 contents
show the value of Headphone jack contsols which return true when plugged
and false when unplugged
control.25 {
iface CARD
name 'Headphone Jack'
value false
refer to the user maunal , the headphone not at the middle is the combo
jack which can be used for S/PDIF
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Title:
N90SV, Realtek ALC663, Mic,
you need to add --log-time=true when starting pulseaudio
I: [pulseaudio] source-output.c: application.icon_name = audacity
I: [pulseaudio] source-output.c: module-stream-restore.id =
source-output-by-application-name:ALSA plug-in [audacity]
D: [pulseaudio] memblockq.c: memblockq
The pincap of node 0x1e support DETECT
Node 0x1e [Pin Complex] wcaps 0x400780: Mono Digital
Control: name=SPDIF Phantom Jack, index=0, device=0
Pincap 0x0014: OUT Detect
Pin Default 0x99430140: [Fixed] SPDIF Out at Int ATAPI
Conn = ATAPI, Color = Unknown
DefAssociation = 0x4,
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=ea9b43addc4d90ca5b029f47f85ca152320a1e8d
!!PCI Soundcards installed in the system
!!--
00:0f.0 Audio device: Silicon Integrated Systems [SiS] Azalia Audio
Controller
!!Aplay/Arecord
** Attachment added: audacity.log
https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1259070/+attachment/3927496/+files/audacity.log
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** Attachment added: test-audacity.ogg
https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1259070/+attachment/3927499/+files/test-audacity.ogg
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** Attachment added: pulseverbose.log
https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1259070/+attachment/3927497/+files/pulseverbose.log
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** Attachment added: Recording with audacity, input set to pulse
https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1259070/+attachment/3927501/+files/tstaudacity.ogg
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** Attachment added: test-arecord.ogg
https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1259070/+attachment/3927498/+files/test-arecord.ogg
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** Attachment added: pulsetest.ogg
https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1259070/+attachment/3927495/+files/pulsetest.ogg
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O.k., so some more testing. Here's what I did:
$ pulseaudio -k LANG=C pulseaudio -vvv ~/pulseverbose.log 21
$ parecord -r pulsetest.wav
record sample in audacity and export to test-audacity.wav
$ pulseaudio -k LANG=C pulseaudio -vvv ~/arecord.log 21
arecord -f cd -d 5 test.wav
pulseaudio -k
APLAY
List of PLAYBACK Hardware Devices
card 0: SIS966 [HDA SIS966], device 0: ALC663 Analog [ALC663 Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: SIS966 [HDA SIS966], device 3: ALC663 Digital [ALC663 Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0
seem lost
does the red light in the optical spdif turn on/off if you toggle IEC958
playback switch
Simple mixer control 'IEC958',0
Capabilities: pswitch pswitch-joined
Playback channels: Mono
Mono: Playback [off]
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I had the probe=mask1 disabled and rebooted before doing the recordings,
so this is standard setup right now.
I don't have any hdmi screens, so can't test that. I also don't have any spdif
devices either. I don't see any lights turning on when switching to hdmi or
analog surround output which I
if your NVIDIA® GeForce® GT 130M does not have audio codec , the
secondary SPDIF-OUT output need to be used for HDMI
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Title:
N90SV, Realtek
did you toggle the IEC958 Playback Switch ?
can you post the output of alsa-info.sh
https://wiki.ubuntu.com/Audio/AlsaInfo
http://www.realtek.com/products/productsView.aspx?Langid=1PNid=24PFid=37Level=5Conn=4ProdID=165
The ALC663 is a 5.1 Channel High Definition Audio Codec
The ALC663
Funnily, parecord gives better output than arecord, audacity using
pulse input is unbearable, but using alsa hw0:0 seems fine again.
do you mean stereo recording using hw:0,0 is OK ?
arecord -c 2 -Dhw:0,0 stereo.wav
how about mono recording ?
arecord -c 1 -Dplughw:0,0 mono.wav
the signal
snd_hda_intel: model=auto probe_mask=1
any specific reason to use probe_mask ? do you want to disable another
codec ?
do it help if you mute either left or right channel ?
Simple mixer control 'Capture',0
Capabilities: cvolume cswitch
Capture channels: Front Left - Front Right
Limits:
Well, I had looked on the internet for solutions and found some like
model=asus-mode1 etc.
I tried most of those but nothing helped. The model=auto probe_mask=1 is the
last one I tried and as it didn't make any difference, I haven't deleted it
(yet).
Muting one channel does not help, trying to
One more: I had to put
pci=nomsi acpi_osi=linux nox2apic
as kernel options, otherwise the system wouldn't even start. Don't know whether
there is any connection...
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but synthom_pulseaudio.log did not contain info about the capture since
only alsa-sink was logged
do you mean directly using alsa work well ?
arecord -Dhw:0,0 -c2 test.wav
you have to provide pulseaudio verbose log while you capture
https://wiki.ubuntu.com/PulseAudio/Log
** Changed in:
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