Bad VoIP performance cause
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Vít Šesták
May 22, 2015, 5:08:51 PM5/22/15
to qubes...@googlegroups.com
Hello,
I am trying to find why I am not able to make any VoIP calls from Qubes. Sound recording and playing works correctly. Both UDP (measured by iperf) and TCP bandwidth is OK, see:
% iperf -c v6ak.com -u -f m -b 10M
------------------------------------------------------------
Client connecting to v6ak.com, UDP port 5001
Sending 1470 byte datagrams
UDP buffer size: 0.20 MByte (default)
------------------------------------------------------------
[ 3] local 10.137.1.23 port 38015 connected with 46.28.110.250 port 5001
[ ID] Interval Transfer Bandwidth
[ 3] 0.0-10.0 sec 11.9 MBytes 10.0 Mbits/sec
[ 3] Sent 8505 datagrams
[ 3] Server Report:
[ 3] 0.0-10.0 sec 11.8 MBytes 9.88 Mbits/sec 0.589 ms 107/ 8505 (1.3%)le jitter
Other parameters look also good on benchmarks.
I can call from the network using my tablet, so network quality is presumably good enough. Moreover, it used to be OK with bare-metal Ubuntu with the same hardware, the same VoIP software and the same network.
But when I make a call using Linphone (SIP), Firefox Hello (WebRTC) or Skype, I get unusable sound quality – almost no sound. Linphone has pointed me to some issue that was later getter visualised by gnome-system-monitor. There is almost no bandwith used and it regurarly goes to zero (or almost zero). See the attachment. This is in contrast with the bandwidth of iperf.
So, where does the issue lie?
a) Bad app. But I've tested multiple different apps that used to work on bare-metal Ubuntu or even on virtualized Ubuntu.
b) Insufficient network bandwidth. The iperf does not confirm it.
c) Issues with VoIP provider. I've tried multiple providers, none of them worked. Moreover, on my tablet connected to the same network, SIP works fine.ther idea
d) Some issues with latency or jitter. First, such issue would not result in low bandwidth used. Second, some measurements suggests good values like 12ms latency and little (I don't remember) jitter.
e) High packet loss. First, such issue would likely be reflected in iperf results. Second, such issue would not show low outbound bandwidth from the same VM, unless the packets are lost inside the VM, which is not plausible…
f) Any other idea?
Regards,
Vít Šesták 'v6ak'
I am trying to find why I am not able to make any VoIP calls from Qubes. Sound recording and playing works correctly. Both UDP (measured by iperf) and TCP bandwidth is OK, see:
% iperf -c v6ak.com -u -f m -b 10M
------------------------------------------------------------
Client connecting to v6ak.com, UDP port 5001
Sending 1470 byte datagrams
UDP buffer size: 0.20 MByte (default)
------------------------------------------------------------
[ 3] local 10.137.1.23 port 38015 connected with 46.28.110.250 port 5001
[ ID] Interval Transfer Bandwidth
[ 3] 0.0-10.0 sec 11.9 MBytes 10.0 Mbits/sec
[ 3] Sent 8505 datagrams
[ 3] Server Report:
[ 3] 0.0-10.0 sec 11.8 MBytes 9.88 Mbits/sec 0.589 ms 107/ 8505 (1.3%)le jitter
Other parameters look also good on benchmarks.
I can call from the network using my tablet, so network quality is presumably good enough. Moreover, it used to be OK with bare-metal Ubuntu with the same hardware, the same VoIP software and the same network.
But when I make a call using Linphone (SIP), Firefox Hello (WebRTC) or Skype, I get unusable sound quality – almost no sound. Linphone has pointed me to some issue that was later getter visualised by gnome-system-monitor. There is almost no bandwith used and it regurarly goes to zero (or almost zero). See the attachment. This is in contrast with the bandwidth of iperf.
So, where does the issue lie?
a) Bad app. But I've tested multiple different apps that used to work on bare-metal Ubuntu or even on virtualized Ubuntu.
b) Insufficient network bandwidth. The iperf does not confirm it.
c) Issues with VoIP provider. I've tried multiple providers, none of them worked. Moreover, on my tablet connected to the same network, SIP works fine.ther idea
d) Some issues with latency or jitter. First, such issue would not result in low bandwidth used. Second, some measurements suggests good values like 12ms latency and little (I don't remember) jitter.
e) High packet loss. First, such issue would likely be reflected in iperf results. Second, such issue would not show low outbound bandwidth from the same VM, unless the packets are lost inside the VM, which is not plausible…
f) Any other idea?
Regards,
Vít Šesták 'v6ak'
Marek Marczykowski-Górecki
May 22, 2015, 5:44:30 PM5/22/15
to Vít Šesták, qubes...@googlegroups.com
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Did you tried "audio low latency" mode[1]? VM have no direct influence
on audio latency, which means that sound played by the VM can hit the
speakers with any delay (some pulseaudio/driver default), not known to
the VM. The same applies for sound recording (I assume you've enabled it
for this VM, right?) - when VM starts recording, the first sample can
arrive with some delay - and this can be the source of your problems.
"audio_low_latency" option tries to force pulseaudio in dom0 to use as
little delay as possible.
[1] https://www.qubes-os.org/doc/UserDoc/ConfigFiles/
- --
Best Regards,
Marek Marczykowski-Górecki
Invisible Things Lab
A: Because it messes up the order in which people normally read text.
Q: Why is top-posting such a bad thing?
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on audio latency, which means that sound played by the VM can hit the
speakers with any delay (some pulseaudio/driver default), not known to
the VM. The same applies for sound recording (I assume you've enabled it
for this VM, right?) - when VM starts recording, the first sample can
arrive with some delay - and this can be the source of your problems.
"audio_low_latency" option tries to force pulseaudio in dom0 to use as
little delay as possible.
[1] https://www.qubes-os.org/doc/UserDoc/ConfigFiles/
- --
Best Regards,
Marek Marczykowski-Górecki
Invisible Things Lab
A: Because it messes up the order in which people normally read text.
Q: Why is top-posting such a bad thing?
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Franz
May 22, 2015, 6:02:20 PM5/22/15
to Marek Marczykowski-Górecki, Vít Šesták, qubes...@googlegroups.com
This solution is described to involve much higher CPU usage in dom0. But when does this higher usage happen? All the time or only when a microphone is used? It would be interesting to have the low latency option (and high CPU usage) only when a microphone is attached to a VM. Even better: the simple fact of attaching a microphone to a VM would activate the low latency option.
In any case it is possible to avoid high CPU usage using a cheap external USB audio card see this for more informations:
Best
Frans only when some
- --
Best Regards,
Marek Marczykowski-Górecki
Invisible Things Lab
A: Because it messes up the order in which people normally read text.
Q: Why is top-posting such a bad thing?
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Marek Marczykowski-Górecki
May 22, 2015, 6:15:45 PM5/22/15
to Franz, Vít Šesták, qubes...@googlegroups.com
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- --
Best Regards,
Marek Marczykowski-Górecki
Invisible Things Lab
A: Because it messes up the order in which people normally read text.
Q: Why is top-posting such a bad thing?
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Franz
May 22, 2015, 6:55:52 PM5/22/15
to Marek Marczykowski-Górecki, Vít Šesták, qubes...@googlegroups.com
So this is not a big issue. Interesting! Thanks Marek
Vít Šesták
May 23, 2015, 3:49:08 AM5/23/15
to qubes...@googlegroups.com, marm...@invisiblethingslab.com, groups-no-private-mail--con...@v6ak.com
What do I need restart after editing guid.conf? When I do a syntax error, I can't even start a VM, which suggests that restarting the VM is enough. Is that correct?
So, after VM reboot, I still can play Youtube videos, but it is all that work. Microphone does not seem to work at all. For example:
The gnome-sound-recorder records a very short noise (about 1s after about 10s runs isning) and even the time elapsed does not look OK.
Just running `arecord -N -f cd -t raw | time head -c1` shows there is significant delay before first byte received from microphone:
Recording raw data 'stdin' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
�head -c1 0.00s user 0.01s system 0% cpu 12.119 total
^[[1;5D^CAborted by signal Interrupt...
It gives me very variable result. Maybe the CPU can't process it in real time? Is Intel(R) Core(TM) i7-2670QM CPU @ 2.20GHz too slow for that?
Command `arecord -t raw | aplay` gives me some feedback loop (low quality and with some underruns) form microphone to my headphones when low latency audio is *disabled*, but it gives me just some short noise shots when low latency audio is enabled.
Regards,
Vít Šesták 'v6ak'
So, after VM reboot, I still can play Youtube videos, but it is all that work. Microphone does not seem to work at all. For example:
The gnome-sound-recorder records a very short noise (about 1s after about 10s runs isning) and even the time elapsed does not look OK.
Just running `arecord -N -f cd -t raw | time head -c1` shows there is significant delay before first byte received from microphone:
Recording raw data 'stdin' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
�head -c1 0.00s user 0.01s system 0% cpu 12.119 total
^[[1;5D^CAborted by signal Interrupt...
It gives me very variable result. Maybe the CPU can't process it in real time? Is Intel(R) Core(TM) i7-2670QM CPU @ 2.20GHz too slow for that?
Command `arecord -t raw | aplay` gives me some feedback loop (low quality and with some underruns) form microphone to my headphones when low latency audio is *disabled*, but it gives me just some short noise shots when low latency audio is enabled.
Regards,
Vít Šesták 'v6ak'
Marek Marczykowski-Górecki
May 23, 2015, 7:03:44 AM5/23/15
to Vít Šesták, qubes...@googlegroups.com
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On Sat, May 23, 2015 at 12:49:08AM -0700, Vít Šesták wrote:
> What do I need restart after editing guid.conf? When I do a syntax error, I
> can't even start a VM, which suggests that restarting the VM is enough. Is
> that correct?
Yes.
> What do I need restart after editing guid.conf? When I do a syntax error, I
> can't even start a VM, which suggests that restarting the VM is enough. Is
> that correct?
> So, after VM reboot, I still can play Youtube videos, but it is all that
> work. Microphone does not seem to work at all. For example:
related before 3.0.5.
> The gnome-sound-recorder records a very short noise (about 1s after about
> 10s runs isning) and even the time elapsed does not look OK.
>
> Just running `arecord -N -f cd -t raw | time head -c1` shows there is
> significant delay before first byte received from microphone:
> Recording raw data 'stdin' : Signed 16 bit Little Endian, Rate 44100 Hz,
> Stereo
> ^[[1;5D^CAborted by signal Interrupt...
Ok, something is wrong here. I've even managed to not receive the data
at all (at least not within about 2 minutes...).
But, when I start any sound application in dom0 (for example
pavucontrol), it works instantly. Looking at pacat log
(/var/log/qubes/pacat.VMNAME.log) it may be something about device
automatic suspend.
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Vít Šesták
May 23, 2015, 8:25:10 AM5/23/15
to qubes...@googlegroups.com, groups-no-private-mail--con...@v6ak.com
What do you mean by qubes-gui-vm? Is that qubes-gui-agent? I've that package in version 3.0.5+jessie1 . If you mean qubes-gui-common, I've it in version 3.0.2+jessie1.
Marek Marczykowski-Górecki
May 23, 2015, 8:34:02 AM5/23/15
to Vít Šesták, qubes...@googlegroups.com
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On Sat, May 23, 2015 at 05:25:10AM -0700, Vít Šesták wrote:
> What do you mean by qubes-gui-vm? Is that qubes-gui-agent? I've that
> package in version 3.0.5+jessie1 . If you mean qubes-gui-common, I've it in
> version 3.0.2+jessie1.
Ah, debian. That's qubes-gui-agent. Anyway there is clearly something
> What do you mean by qubes-gui-vm? Is that qubes-gui-agent? I've that
> package in version 3.0.5+jessie1 . If you mean qubes-gui-common, I've it in
> version 3.0.2+jessie1.
wrong, with either side of audio connection. I'm trying to find out
what.
- --
Best Regards,
Marek Marczykowski-Górecki
Invisible Things Lab
A: Because it messes up the order in which people normally read text.
Q: Why is top-posting such a bad thing?
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Marek Marczykowski-Górecki
May 23, 2015, 9:37:18 PM5/23/15
to Vít Šesták, qubes...@googlegroups.com
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On Sat, May 23, 2015 at 02:33:53PM +0200, Marek Marczykowski-Górecki wrote:
> On Sat, May 23, 2015 at 05:25:10AM -0700, Vít Šesták wrote:
> > What do you mean by qubes-gui-vm? Is that qubes-gui-agent? I've that
> > package in version 3.0.5+jessie1 . If you mean qubes-gui-common, I've it in
> > version 3.0.2+jessie1.
>
> Ah, debian. That's qubes-gui-agent. Anyway there is clearly something
> wrong, with either side of audio connection. I'm trying to find out
> what.
Stream parameters in low latency mode were invalid for recording.
> On Sat, May 23, 2015 at 05:25:10AM -0700, Vít Šesták wrote:
> > What do you mean by qubes-gui-vm? Is that qubes-gui-agent? I've that
> > package in version 3.0.5+jessie1 . If you mean qubes-gui-common, I've it in
> > version 3.0.2+jessie1.
>
> Ah, debian. That's qubes-gui-agent. Anyway there is clearly something
> wrong, with either side of audio connection. I'm trying to find out
> what.
I've fixed that, and also added flag PA_STREAM_ADJUST_LATENCY, which
should lower latency even further. It can cause buffer underruns if your
VM do not send the data fast enough, so I'm not sure if this should stay
there.
The code:
https://github.com/marmarek/qubes-gui-daemon/tree/audio-fixes
I've also uploaded test package to unstable repository, you can install
it with:
sudo qubes-dom0-update --enablerepo=qubes-dom0-unstable qubes-gui-dom0
You should get qubes-gui-dom0-3.0.4-1.1.fc20.x86_64
Please let me know how is it working for you.
To get back to previous version (assuming 3.0.4 from current-testing), call:
sudo qubes-dom0-update --clean --action=downgrade
- --enablerepo=qubes-dom0-current-testing
qubes-gui-dom0-3.0.4-1.fc20.x86_64
- --
Best Regards,
Marek Marczykowski-Górecki
Invisible Things Lab
A: Because it messes up the order in which people normally read text.
Q: Why is top-posting such a bad thing?
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Vít Šesták
May 24, 2015, 8:45:38 AM5/24/15
to qubes...@googlegroups.com, groups-no-private-mail--con...@v6ak.com
Thank you for the update, but I am afraid I can't install it. It seemingly does not know about the new version:
https://gist.github.com/v6ak/061edd321549c9e098d2
Regards,
Vít Šesták 'v6ak'
https://gist.github.com/v6ak/061edd321549c9e098d2
Regards,
Vít Šesták 'v6ak'
Marek Marczykowski-Górecki
May 24, 2015, 12:59:46 PM5/24/15
to Vít Šesták, qubes...@googlegroups.com
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On Sun, May 24, 2015 at 05:45:38AM -0700, Vít Šesták wrote:
> Thank you for the update, but I am afraid I can't install it. It seemingly
> does not know about the new version:
> https://gist.github.com/v6ak/061edd321549c9e098d2
Try now.
> Thank you for the update, but I am afraid I can't install it. It seemingly
> does not know about the new version:
> https://gist.github.com/v6ak/061edd321549c9e098d2
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Vít Šesták
May 24, 2015, 4:11:37 PM5/24/15
to qubes...@googlegroups.com, groups-no-private-mail--con...@v6ak.com
The installation works now, but I'll reboot Qubes and test it probably tomorrow.
v6ak
v6ak
Vít Šesták
May 25, 2015, 6:05:59 AM5/25/15
to qubes...@googlegroups.com
OK, I tested it with both Linphone (SIP) and Skype and gnome-sound-recorder.
It definitely works now either with low latency audio or without, although with somehow poor quality. It sounds like a low bitrate, even with gnome-sound-recorder, which should exclude network issues. (I am not saying that it is a low bitrate, I am just describing how it sounds like.) With LLA, the quality is subjectively slightly better. Even with LLA, I don't see significant Dom0 CPU usage.
Thank you for the improvements!
Regards,
Vít Šesták 'v6ak'
It definitely works now either with low latency audio or without, although with somehow poor quality. It sounds like a low bitrate, even with gnome-sound-recorder, which should exclude network issues. (I am not saying that it is a low bitrate, I am just describing how it sounds like.) With LLA, the quality is subjectively slightly better. Even with LLA, I don't see significant Dom0 CPU usage.
Thank you for the improvements!
Regards,
Vít Šesták 'v6ak'
Marek Marczykowski-Górecki
May 25, 2015, 6:03:10 PM5/25/15
to Vít Šesták, qubes...@googlegroups.com
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On Mon, May 25, 2015 at 03:05:59AM -0700, Vít Šesták wrote:
> OK, I tested it with both Linphone (SIP) and Skype and gnome-sound-recorder.
>
> It definitely works now either with low latency audio or without, although
> with somehow poor quality. It sounds like a low bitrate, even with
> gnome-sound-recorder, which should exclude network issues. (I am not saying
> that it is a low bitrate, I am just describing how it sounds like.) With
> LLA, the quality is subjectively slightly better. Even with LLA, I don't
> see significant Dom0 CPU usage.
I guess that high CPU usage was actually because of just fixed bug.
> OK, I tested it with both Linphone (SIP) and Skype and gnome-sound-recorder.
>
> It definitely works now either with low latency audio or without, although
> with somehow poor quality. It sounds like a low bitrate, even with
> gnome-sound-recorder, which should exclude network issues. (I am not saying
> that it is a low bitrate, I am just describing how it sounds like.) With
> LLA, the quality is subjectively slightly better. Even with LLA, I don't
> see significant Dom0 CPU usage.
Previously it copied audio stream one sample at a time, so there was a
lot function calls (44.1kHz...).
Check /var/log/qubes/pacat.VMNAME.log in dom0 - there you may find some
messages about stream underruns.
The only thinks I can think of which can affect quality here are:
- delay of samples (jitter)
- implied resample (Qubes forces 44.1kHz, signed 16bit LE)
This is no compression or any other explicit stream transformation.
In case of sample delay, I guess it would affect real time streaming, but
not recording...
Maybe this is something about your hardware or drivers? Can you check
the same in dom0 (parecord + paplay) and compare? Try with default
settings and later with --format=s16le --rate=44100 --channels=2.
> Thank you for the improvements!
- --
Best Regards,
Marek Marczykowski-Górecki
Invisible Things Lab
A: Because it messes up the order in which people normally read text.
Q: Why is top-posting such a bad thing?
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Best Regards,
Marek Marczykowski-Górecki
Invisible Things Lab
A: Because it messes up the order in which people normally read text.
Q: Why is top-posting such a bad thing?
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Vít Šesták
May 26, 2015, 4:36:03 AM5/26/15
to qubes...@googlegroups.com, groups-no-private-mail--con...@v6ak.com
Great news! So the LLA option might be defaulted to true or maybe even removed in future, if it will work correctly for others :)
I originally tried to record in dom0, but arecord reported the device to be busy, so I thought that the recording is somehow exclusively connected to some daemon. The dom0:parecord pointed me to the last issue and I was able to resolve it. It gave me similarly poor sound, but with some loud noise. The remaining issue was just in my audio settings: The poor quality was likely caused by reaching some volume limit. The fix was simple:
In dom0:xfce4-mixer, I lowered the volume (note that I have two audio sources, so it is important to lower the correct one). I also disabled the external microphone, but that's probably unneeded, as the system correctly detects it is unplugged. In KMix, I increased the recording volume for the VM, as it was low. The sound quality became good then.
There was one more issue that I fixed almost immediatelly, so I didn't mention it. But someone might find it useful. I was getting some noise even in the VM. The noise was brought from “Mic boost” in “HCA Intel PCH (Alsa mixer)”. (Note that this is in “Playback” in xfce4-mixer, which might be confusing.) I turned it to zero. Well, as there is also “Internal Mic boost”, which is non-zero, the noise was probably brought by the unplugged external microphone, so this might be unneeded after disabling the external microphone. Note that this is not Qubes-specific issue, I had the same issue with Ubuntu 12.04 on the same machine.
In Qubes, I am, however, slightly confused by audio settings. In Ubuntu, I used gnome-alsamixer for everything. In Qubes, I use mostly KMix, but I sometimes have to use xfce4-mixer (e.g. for configuring the mic boost) or maybe pavucontrol. That's not a big issue for me, just slightly uncomfortable, as I don't have to do that every day.
Thanks for your fixes and assistance!
Regards,
Vít Šesták 'v6ak'
I originally tried to record in dom0, but arecord reported the device to be busy, so I thought that the recording is somehow exclusively connected to some daemon. The dom0:parecord pointed me to the last issue and I was able to resolve it. It gave me similarly poor sound, but with some loud noise. The remaining issue was just in my audio settings: The poor quality was likely caused by reaching some volume limit. The fix was simple:
In dom0:xfce4-mixer, I lowered the volume (note that I have two audio sources, so it is important to lower the correct one). I also disabled the external microphone, but that's probably unneeded, as the system correctly detects it is unplugged. In KMix, I increased the recording volume for the VM, as it was low. The sound quality became good then.
There was one more issue that I fixed almost immediatelly, so I didn't mention it. But someone might find it useful. I was getting some noise even in the VM. The noise was brought from “Mic boost” in “HCA Intel PCH (Alsa mixer)”. (Note that this is in “Playback” in xfce4-mixer, which might be confusing.) I turned it to zero. Well, as there is also “Internal Mic boost”, which is non-zero, the noise was probably brought by the unplugged external microphone, so this might be unneeded after disabling the external microphone. Note that this is not Qubes-specific issue, I had the same issue with Ubuntu 12.04 on the same machine.
In Qubes, I am, however, slightly confused by audio settings. In Ubuntu, I used gnome-alsamixer for everything. In Qubes, I use mostly KMix, but I sometimes have to use xfce4-mixer (e.g. for configuring the mic boost) or maybe pavucontrol. That's not a big issue for me, just slightly uncomfortable, as I don't have to do that every day.
Thanks for your fixes and assistance!
Regards,
Vít Šesták 'v6ak'
Franz
May 26, 2015, 12:18:49 PM5/26/15
to Vít Šesták, qubes...@googlegroups.com
So resuming Marek made some fixes in R3 (will not work in R2) that solve very long term audio issues without involving high CPU usage. Also to get good audio quality some tweaks are still required as reported in last Vit email. Is that correct?
If this is correct it would be really important if Vit could prepare an easier tutorial for these tweaks to put it in the Qubes wiki for everybody to follow. This is not some rare need of some venture project. This is a very basic and common use of a laptop. Having solved it is a major milestone and I congratulate.
Best
Fran
Regards,
Vít Šesták 'v6ak'
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Vít Šesták
May 27, 2015, 5:20:15 AM5/27/15
to qubes...@googlegroups.com, groups-no-private-mail--con...@v6ak.com
OK, I'll summarize that. Note that I am using 3.0-RC1 and I've never tried any older Qubes release.
First, Marek has fixed the issue in new version. I am not sure if the fix has been delivered to the stable repo and I have no idea how will be the R2 supported and if the fix will be or was backported or if there will be no fix for that.
Second, you can enable low latency audio, which seems to have little or no drawbacks in the new version. I hope this will be the default option or even the only option in future, but that's just my guess and not an official announcement.
Third, I've realized something about audio settings. You may be fine without these finding, especially if you haven't experimented with audio settings and your hardware is supported well. It might be useful to read them, especially the first one, if you want to prevent issues:
Finding #1: There are various tools for configuring audio volume, e.g. xfce4-mixer, KMix and pavucontrol. They are different tools and they support setting different parameters. It seems to be wise to use only one of these tools unless you really need to use another. If you use more of them, you might configure something that is not configurable in the other tool and then wonder why the sound is off or so. Although this is trivial to fix (just use the other tool once more and configure the parameter you need), it might not be apparent at first sight in some cases.
I've configured the sound settings only from dom0 and not from AppVMs, which seems to be wise (unless you actually need to configure something from an AppVM). It would introduce even more complexity in sound settings.
Don't take the advice(s) in finding #1 too strictly. You may need not to follow the advice in some special cases (see below). If you know what you have changed in what VM and what volume control app, you probably can easily revert these changes if you need.
Finding #2: It does not sound like a good idea to configure global volume to be high and per-VM volume to be low (or even vice-versa), as it seemingly causes some rounding inaccuracies and a poor sound quality. It might sound like low bitrate, even if the bitrate is not so low. You probably will not get such issue if you follow advice in finding #1. I had configured the microphone for maximum recording volume globally (you can configure that in xfce4-mixer, but you probably can't configure that in KMix) and just low volume per-VM (you can configure it in KMix, but I am unsure about xfce4-mixer). I've solved this by setting a medium recording volume for both.
Finding #3: I had some issue with noise in sound input. If you are lucky, you don't need to read this paragraph. The issue is not Qubes-specific, but more likely hardware specific, as I had the issue also on bare-metal Ubuntu 12.04. If it wasn't HW-specific issue, it would probably have been fixed. My laptop has an internal microphone and a jack for external microphone (which I've never used). The noise disappeared when I turned “Mic boost” in “HCA Intel PCH (Alsa mixer)” to zero. (Note that this is in “Playback” in xfce4-mixer, which might be confusing. I haven't find this config in KMix, so this was the time for breaking the advice I gave in my Finding #1…) It was likely some noise from the external microphone jack connector, which I have never used.
Regards,
Vít Šesták 'v6ak'
First, Marek has fixed the issue in new version. I am not sure if the fix has been delivered to the stable repo and I have no idea how will be the R2 supported and if the fix will be or was backported or if there will be no fix for that.
Second, you can enable low latency audio, which seems to have little or no drawbacks in the new version. I hope this will be the default option or even the only option in future, but that's just my guess and not an official announcement.
Third, I've realized something about audio settings. You may be fine without these finding, especially if you haven't experimented with audio settings and your hardware is supported well. It might be useful to read them, especially the first one, if you want to prevent issues:
Finding #1: There are various tools for configuring audio volume, e.g. xfce4-mixer, KMix and pavucontrol. They are different tools and they support setting different parameters. It seems to be wise to use only one of these tools unless you really need to use another. If you use more of them, you might configure something that is not configurable in the other tool and then wonder why the sound is off or so. Although this is trivial to fix (just use the other tool once more and configure the parameter you need), it might not be apparent at first sight in some cases.
I've configured the sound settings only from dom0 and not from AppVMs, which seems to be wise (unless you actually need to configure something from an AppVM). It would introduce even more complexity in sound settings.
Don't take the advice(s) in finding #1 too strictly. You may need not to follow the advice in some special cases (see below). If you know what you have changed in what VM and what volume control app, you probably can easily revert these changes if you need.
Finding #2: It does not sound like a good idea to configure global volume to be high and per-VM volume to be low (or even vice-versa), as it seemingly causes some rounding inaccuracies and a poor sound quality. It might sound like low bitrate, even if the bitrate is not so low. You probably will not get such issue if you follow advice in finding #1. I had configured the microphone for maximum recording volume globally (you can configure that in xfce4-mixer, but you probably can't configure that in KMix) and just low volume per-VM (you can configure it in KMix, but I am unsure about xfce4-mixer). I've solved this by setting a medium recording volume for both.
Finding #3: I had some issue with noise in sound input. If you are lucky, you don't need to read this paragraph. The issue is not Qubes-specific, but more likely hardware specific, as I had the issue also on bare-metal Ubuntu 12.04. If it wasn't HW-specific issue, it would probably have been fixed. My laptop has an internal microphone and a jack for external microphone (which I've never used). The noise disappeared when I turned “Mic boost” in “HCA Intel PCH (Alsa mixer)” to zero. (Note that this is in “Playback” in xfce4-mixer, which might be confusing. I haven't find this config in KMix, so this was the time for breaking the advice I gave in my Finding #1…) It was likely some noise from the external microphone jack connector, which I have never used.
Regards,
Vít Šesták 'v6ak'
Franz
May 27, 2015, 8:16:48 AM5/27/15
to Vít Šesták, qubes...@googlegroups.com
Many thanks. Now I cannot test it because am running R2. But plan to move to R3.
Best
Fran
Regards,--
Vít Šesták 'v6ak'
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Marek Marczykowski-Górecki
May 27, 2015, 1:59:09 PM5/27/15
to Vít Šesták, qubes...@googlegroups.com
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hash: SHA1
On Wed, May 27, 2015 at 02:20:15AM -0700, Vít Šesták wrote:
> OK, I'll summarize that. Note that I am using 3.0-RC1 and I've never tried
> any older Qubes release.
>
> First, Marek has fixed the issue in new version. I am not sure if the fix
> has been delivered to the stable repo and I have no idea how will be the R2
> supported and if the fix will be or was backported or if there will be no
> fix for that.
I'll backport those fixes to R2 soon. But the final result could be
> OK, I'll summarize that. Note that I am using 3.0-RC1 and I've never tried
> any older Qubes release.
>
> First, Marek has fixed the issue in new version. I am not sure if the fix
> has been delivered to the stable repo and I have no idea how will be the R2
> supported and if the fix will be or was backported or if there will be no
> fix for that.
different, because of different (lower performance) vchan
implementation.
> Second, you can enable low latency audio, which seems to have little or no
> drawbacks in the new version. I hope this will be the default option or
> even the only option in future, but that's just my guess and not an
> official announcement.
will not reveal any drawback, this will probably change in R3.1.
- --
Best Regards,
Marek Marczykowski-Górecki
Invisible Things Lab
A: Because it messes up the order in which people normally read text.
Q: Why is top-posting such a bad thing?
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