N90SV, Realtek ALC663, Mic, Internal Underruns, dropouts or crackling sound

[HinzundKunz]

Recording directly with arecord works, but using pulseaudio has crackling noise.

Workaround is to disable timer based scheduling in /etc/pulse/default.pa
AND setting

default-fragments = 4
default-fragment-size-msec = 90 (or higher, lower still produces noise)

in daemon.conf. Total hw buffer size is 371 ms (according to pulseaudio -vvv > pulse.log)

ProblemType: Bug
DistroRelease: Ubuntu 13.10
Package: alsa-base 1.0.25+dfsg-0ubuntu4
ProcVersionSignature: Ubuntu 3.11.0-15.22-generic 3.11.10
Uname: Linux 3.11.0-15-generic i686
ApportVersion: 2.12.5-0ubuntu2.1
Architecture: i386
AudioDevicesInUse:
 USER PID ACCESS COMMAND
 /dev/snd/controlC0: martin 2432 F.... pulseaudio
Date: Mon Dec 9 09:29:29 2013
InstallationDate: Installed on 2013-04-22 (230 days ago)
InstallationMedia: Kubuntu 12.10 "Quantal Quetzal" - Release i386 (20121017.1)
MarkForUpload: True
PackageArchitecture: all
SourcePackage: alsa-driver
Symptom: audio
Symptom_AlsaRecordingTest: ALSA recording test through plughw:SIS966 failed
Symptom_Card: Internes Audio - HDA SIS966
Symptom_Jack: Mic, Internal
Symptom_Type: Underruns, dropouts, or "crackling" sound
Title: [N90SV, Realtek ALC663, Mic, Internal] Underruns, dropouts or crackling sound
UpgradeStatus: Upgraded to saucy on 2013-10-18 (51 days ago)
dmi.bios.date: 02/25/2010
dmi.bios.vendor: American Megatrends Inc.
dmi.bios.version: 207
dmi.board.asset.tag: ATN12345678901234567
dmi.board.name: N90SV
dmi.board.vendor: ASUS CORPORATION
dmi.board.version: 1.0
dmi.chassis.type: 10
dmi.chassis.vendor: ASUSTeK Computer Inc.
dmi.modalias: dmi:bvnAmericanMegatrendsInc.:bvr207:bd02/25/2010:svnASUSTeKComputerInc.:pnN90SV:pvr1.0:rvnASUSCORPORATION:rnN90SV:rvr1.0:cvnASUSTeKComputerInc.:ct10:cvr:
dmi.product.name: N90SV
dmi.product.version: 1.0
dmi.sys.vendor: ASUSTeK Computer Inc.
mtime.conffile..etc.modprobe.d.alsa.base.conf: 2013-12-08T17:23:33.527378

[Raymond (superquad-vortex2)]

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: Capture 0 - 31
  Front Left: Capture 15 [48%] [6.00dB] [on]
  Front Right: Capture 15 [48%] [6.00dB] [on]

[ 145.516392] perf samples too long (2517 > 2500), lowering kernel.perf_event_max_sample_rate to 50000
[ 287.204040] perf samples too long (5005 > 5000), lowering kernel.perf_event_max_sample_rate to 25000

https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1246216

[HinzundKunz (martin-tlustos)]

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 record in mono (in audacity) does not help (and wouldn't work in skype anyway). Recording from line-in has the same noise, too.

Direct recording with arecord does work, though.

[HinzundKunz (martin-tlustos)]

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...

[Raymond (superquad-vortex2)]

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

[HinzundKunz (martin-tlustos)]

O.k., so some more testing. Here's what I did:
$ pulseaudio -k && LANG=C pulseaudio -vvv > ~/pulseverbose.log 2>&1
$ parecord -r pulsetest.wav
record sample in audacity and export to "test-audacity.wav"
$ pulseaudio -k && LANG=C pulseaudio -vvv > ~/arecord.log 2>&1
arecord -f cd -d 5 test.wav
pulseaudio -k && LANG=C pulseaudio -vvv > ~/audacity.log 2>&1
start audacity and record with input hw:0,0 (not pulse)

then converted the output wav files to ogg for size reasons. Find all the files attached.

Funnily, parecord gives better output than arecord, audacity using pulse input is unbearable, but using alsa hw0:0 seems fine again...

My final aim is to make skype working. Right now I can't talk to anybody, they just won't understand me because of all the noise...

[Raymond (superquad-vortex2)]

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 the optical spdif device 1

do you find another HDMI codec if you remove probe_mask=1 ?

Node 0x11 [Pin Complex] wcaps 0x400700: Mono Digital
  Control: name="HDMI Phantom Jack", index=0, device=0
  Pincap 0x00000010: OUT
  Pin Default 0x18561130: [Jack] Digital Out at Int HDMI
    Conn = Digital, Color = Black
    DefAssociation = 0x3, Sequence = 0x0
    Misc = NO_PRESENCE
  Pin-ctls: 0x40: OUT
  Power states: D0 D1 D2 D3 EPSS
  Power: setting=D0, actual=D0
  Connection: 1
     0x10

Node 0x1e [Pin Complex] wcaps 0x400780: Mono Digital
  Control: name="SPDIF Phantom Jack", index=0, device=0
  Pincap 0x00000014: OUT Detect
  Pin Default 0x99430140: [Fixed] SPDIF Out at Int ATAPI
    Conn = ATAPI, Color = Unknown
    DefAssociation = 0x4, Sequence = 0x0
    Misc = NO_PRESENCE
  Pin-ctls: 0x40: OUT
  Unsolicited: tag=00, enabled=0
  Power states: D0 D1 D2 D3 EPSS
  Power: setting=D0, actual=D0
  Connection: 1
     0x06

does it support 5.1 by retasking mic as output ?

1 x Microphone-in jack
2 x Headphone-out jack (1 with S/PDIF)
1 x HDMI

[ 25.166284] autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) type:speaker
[ 25.166287] speaker_outs=0 (0x0/0x0/0x0/0x0/0x0)
[ 25.166289] hp_outs=2 (0x21/0x15/0x0/0x0/0x0)
[ 25.166291] mono: mono_out=0x0
[ 25.166292] dig-out=0x11/0x1e
[ 25.166294] inputs:
[ 25.166296] Internal Mic=0x19
[ 25.166298] Mic=0x18

[Raymond (superquad-vortex2)]

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]

[HinzundKunz (martin-tlustos)]

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 suppose is the same as spdif, right?
My spdif jack is right next to the headphone jack and can be used as such as well. That's pretty much all I can tell.

ANything else I can do for further testing? Adding options to alsa-base.conf and retesting?

[Raymond (superquad-vortex2)]

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=1&PNid=24&PFid=37&Level=5&Conn=4&ProdID=165

The ALC663 is a 5.1 Channel High Definition Audio Codec

The ALC663 provides two independent S/PDIF outputs and supports 16/20/24-bit SPDIF output with a sampling rate of up to 192kHz, offering easy connection of PCs to high quality consumer electronic products such as digital decoders and speakers. In addition to the standard (primary) SPDIF output function, the ALC663 features another independent (secondary) SPDIF-OUT output and converters that transport digital audio output to a High Definition Media Interface (HDMI) transmitter (becoming more common in high-end PCs).

[Raymond (superquad-vortex2)]

if your NVIDIA® GeForce® GT 130M does not have audio codec , the secondary SPDIF-OUT output need to be used for HDMI

[Raymond (superquad-vortex2)]

> 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 will be much lower if the left and right channel of mic input are out of phase when using software downmix with (left+right)/2

[Raymond (superquad-vortex2)]

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
  IEC Coding Type: 0x0
  PCM:
    rates [0x5e0]: 44100 48000 88200 96000 192000
    bits [0xe]: 16 20 24
    formats [0x1]: PCM
  Power states: D0 D1 D2 D3 EPSS
  Power: setting=D0, actual=D0

Node 0x10 [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 Playback Switch", index=0, device=0
  Control: name="IEC958 Default PCM Playback Switch", index=0, device=0
  Device: name="ALC663 Digital", type="HDMI", device=3
  Converter: stream=0, channel=0
  Digital:
  Digital category: 0x0
  IEC Coding Type: 0x0
  PCM:
    rates [0x5e0]: 44100 48000 88200 96000 192000
    bits [0xe]: 16 20 24
    formats [0x1]: PCM
  Power states: D0 D1 D2 D3 EPSS
  Power: setting=D0, actual=D0

[HinzundKunz (martin-tlustos)]

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.

[Raymond (superquad-vortex2)]

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
  comment {
   access read
   type BOOLEAN
   count 1
  }
 }
 control.26 {
  iface CARD
  name 'Headphone Jack'
  index 1
  value false
  comment {
   access read
   type BOOLEAN
   count 1
  }
 }

Node 0x15 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out
  Control: name="Headphone Playback Switch", index=1, device=0
    ControlAmp: chs=3, dir=Out, idx=0, ofs=0
  Control: name="Headphone Jack", index=1, device=0
  Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
  Amp-Out vals: [0x00 0x00]
  Pincap 0x0001003c: IN OUT HP EAPD Detect
  EAPD 0x2: EAPD
  Pin Default 0x01211020: [Jack] HP Out at Ext Rear
    Conn = 1/8, Color = Black
    DefAssociation = 0x2, Sequence = 0x0
  Pin-ctls: 0xc0: OUT HP
  Unsolicited: tag=02, enabled=1
  Power states: D0 D1 D2 D3 EPSS
  Power: setting=D0, actual=D0
  Connection: 2
     0x0c 0x0d*

Node 0x21 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out
  Control: name="Headphone Playback Switch", index=0, device=0
    ControlAmp: chs=3, dir=Out, idx=0, ofs=0
  Control: name="Headphone Jack", index=0, device=0
  Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
  Amp-Out vals: [0x00 0x00]
  Pincap 0x0000001c: OUT HP Detect
  Pin Default 0x0121401f: [Jack] HP Out at Ext Rear
    Conn = 1/8, Color = Green
    DefAssociation = 0x1, Sequence = 0xf
  Pin-ctls: 0xc0: OUT HP
  Unsolicited: tag=01, enabled=1
  Power states: D0 D1 D2 D3 EPSS
  Power: setting=D0, actual=D0
  Connection: 3
     0x0c 0x0d* 0x0e

[Raymond (superquad-vortex2)]

refer to the user maunal , the headphone not at the middle is the combo jack which can be used for S/PDIF

[Raymond (superquad-vortex2)]

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 requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
I: [pulseaudio] protocol-native.c: Final latency 60.00 ms = 30.00 ms + 30.00 ms
D: [alsa-source-ALC663 Analog] alsa-source.c: latency set to 30.00ms
D: [alsa-source-ALC663 Analog] alsa-source.c: hwbuf_unused=60244

audacity seem request low latency and set hwbuf_unused = 60244

D: [alsa-source-ALC663 Analog] alsa-source.c: setting avail_min=442
D: [alsa-source-ALC663 Analog] alsa-source.c: latency set to 30.00ms
D: [alsa-source-ALC663 Analog] alsa-source.c: hwbuf_unused=60244
D: [alsa-source-ALC663 Analog] alsa-source.c: setting avail_min=442
D: [alsa-source-ALC663 Analog] alsa-source.c: Requested volume: 0: 55% 1: 55%
D: [alsa-source-ALC663 Analog] alsa-source.c: in dB: 0: -15.59 dB 1: -15.59 dB
D: [alsa-source-ALC663 Analog] alsa-source.c: Got hardware volume: 0: 53% 1: 53%
D: [alsa-source-ALC663 Analog] alsa-source.c: in dB: 0: -16.50 dB 1: -16.50 dB
D: [alsa-source-ALC663 Analog] alsa-source.c: Calculated software volume: 0: 104% 1: 104% (accurate-enough=no)
D: [alsa-source-ALC663 Analog] alsa-source.c: in dB: 0: 0.91 dB 1: 0.91 dB
D: [alsa-source-ALC663 Analog] source.c: Volume not changing
D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_0f.0.
D: [alsa-source-ALC663 Analog] alsa-source.c: hwbuf_unused=0
D: [alsa-source-ALC663 Analog] alsa-source.c: setting avail_min=15502

not sure why hwbuf_unsused was set to zero

D: [alsa-source-ALC663 Analog] alsa-source.c: Requested volume: 0: 55% 1: 55%
D: [alsa-source-ALC663 Analog] alsa-source.c: in dB: 0: -15.59 dB 1: -15.59 dB
D: [alsa-source-ALC663 Analog] alsa-source.c: Got hardware volume: 0: 53% 1: 53%
D: [alsa-source-ALC663 Analog] alsa-source.c: in dB: 0: -16.50 dB 1: -16.50 dB
D: [alsa-source-ALC663 Analog] alsa-source.c: Calculated software volume: 0: 104% 1: 104% (accurate-enough=no)
D: [alsa-source-ALC663 Analog] alsa-source.c: in dB: 0: 0.91 dB 1: 0.91 dB
D: [alsa-source-ALC663 Analog] source.c: Volume not changing
D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_0f.0.analog-stereo becomes idle, timeout in 5 seconds.
D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_0f.0.analog-stereo becomes idle, timeout in 5 seconds.
D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum.
I: [pulseaudio] source-output.c: Freeing output 17 "ALSA Capture"
I: [pulseaudio] module-stream-restore.c: Restoring device for stream source-output-by-application-n...

[Raymond (superquad-vortex2)]

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 0x00000014: OUT Detect
  Pin Default 0x99430140: [Fixed] SPDIF Out at Int ATAPI
    Conn = ATAPI, Color = Unknown
    DefAssociation = 0x4, Sequence = 0x0
    Misc = NO_PRESENCE
  Pin-ctls: 0x40: OUT
  Unsolicited: tag=00, enabled=0
  Power states: D0 D1 D2 D3 EPSS
  Power: setting=D0, actual=D0
  Connection: 1
     0x06

[Raymond (superquad-vortex2)]

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 output
!!--------------------

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

you have to send email to the author

your notebook spec support SPDIF and HDMI

there are only one HDA controller and one HDA codec alc663

codec alc663 have primary and secondary digital output and node 0x1e SPDIF and 0x11 HDMI pin complexes

[HinzundKunz (martin-tlustos)]

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...

[Raymond (superquad-vortex2)]

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

[Raymond (superquad-vortex2)]

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

[Raymond (superquad-vortex2)]

: [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 samples).
I: [alsa-source-ALC663 Analog] alsa-util.c: ALSA period wakeups disabled
D: [alsa-source-ALC663 Analog] alsa-source.c: hwbuf_unused=0
D: [alsa-source-ALC663 Analog] alsa-source.c: setting avail_min=15502
D: [alsa-source-ALC663 Analog] alsa-source.c: hwbuf_unused=0
D: [alsa-source-ALC663 Analog] alsa-source.c: setting avail_min=15502
I: [alsa-source-ALC663 Analog] alsa-source.c: Time scheduling watermark is 20.00ms
I: [alsa-source-ALC663 Analog] alsa-source.c: Resumed successfully...
I: [alsa-source-ALC663 Analog] alsa-source.c: Starting capture.

D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_0f.0.analog-stereo becomes idle, timeout in 5 seconds.
D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
I: [pulseaudio] source-output.c: Created output 17 "ALSA Capture" on alsa_input.pci-0000_00_0f.0.analog-stereo with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: [pulseaudio] source-output.c: media.name = "ALSA Capture"
I: [pulseaudio] source-output.c: application.name = "ALSA plug-in [audacity]"
I: [pulseaudio] source-output.c: native-protocol.peer = "UNIX socket client"
I: [pulseaudio] source-output.c: native-protocol.version = "28"
I: [pulseaudio] source-output.c: application.process.id = "28751"
I: [pulseaudio] source-output.c: application.process.user = "martin"
I: [pulseaudio] source-output.c: application.process.host = "martin-N90SV"
I: [pulseaudio] source-output.c: application.process.binary = "audacity"
I: [pulseaudio] source-output.c: application.language = "de_AT.UTF-8"
I: [pulseaudio] source-output.c: window.x11.display = ":0"
I: [pulseaudio] source-output.c: application.process.machine_id = "b1897efd1271d7193610a64c51750c4a"
I: [pulseaudio] source-output.c: application.process.session_id = "c3"
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 requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
I: [pulseaudio] protocol-native.c: Final latency 60.00 ms = 30.00 ms + 30.00 ms
D: [alsa-source-ALC663 Analog] alsa-source.c: latency set to 30.00ms
D: [alsa-source-ALC663 Analog] alsa-source.c: hwbuf_unused=60244
D: [alsa-source-ALC663 Analog] alsa-source.c: setting avail_min=442
D: [alsa-source-ALC663 Analog] alsa-source.c: latency set to 30.00ms
D: [a...

[HinzundKunz (martin-tlustos)]

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)
Audacity via Alsa Capture:
1411 μs (= buffer: 0 μs + source: 1411 μs)

Both skype and audacity have crackling noise, while parecord doesn't. But if I set --latency-msec to a value below 100msec, it crackles. Above 100msec it barely crackles, and if I set it to 370ms, it does not crackle at all.

[Raymond (superquad-vortex2)]

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 Dump positions on each period update call

 echo 8 > /proc/asound/card0/pcm0c/xrun_debug

[Launchpad Janitor (janitor)]

[Expired for pulseaudio (Ubuntu) because there has been no activity for 60 days.]