LinuxMusicians

First post here, so greets to all,

I have an M-Track Plus, and was able to set up alsa, jack and guitarix to mostly work together on an Arch Linux system. Unfortunately, I'm having periodic xruns no matter what I do. I've tried nearly a hundred different things suggested on this forum and the web, including threadirqs enabled, linux-rt and self-compiled (with config_hz at 1000) kernels, turning off hyper threading and turbo boost in bios, etc. To no avail.

My system is good enough I think (i7-3770, 16GB RAM, SSD, etc) so now I'm wondering if it's the M-Track Plus that might be the problem. I would be grateful to know if anyone here has gotten the M-Track Plus to work on a Linux box.

TIA

Hi there!

It reminds me of a problem I had years ago. The seed of evil was in hardware/kernel interactions involving components on the same bus of the sound-card. I suggest to have a go with trying to disable what you don't need, especially the network. Some info here. I also put a script example here.

Also I uninstall bluetooth and webcam support, and network management software,
and make sure the system is not polling for dvd's, or other media. You might
have some control at the bios level, if there is motherboard hardware
you can live without.

run htop, or top to see if there are processes running that are not music related,
kill them, and remove them from the system auto-start utility.

Search usb in your package manager, and install usb items that are available.

If nothing helps, try some arch spin-offs or older versions with different generation
kernels. Try the avlinux live dvd. If it works, compare configs, libs, kernel modules
loaded etc

Cheers

Awesome guys, thanks for responding.

I will follow your suggestions, and in fact I was already in the process of downloading avlinux just after making my initial post. I will report back on how things go.

By the way, I am running Arch Linux on my new laptop (check my blog for details). I still have to maximize the performances but it is working very good already following these steps (already linked to you before). Maybe you could skim through and check whether there is something you missed. Further potential issues on the top of my head:

Power management software: use tlp.
Is pam installed?
USB ports: are you plugging your device to USB 2 or USB 3? Use USB 2 ports for the sound-cards.

glowrak guy and CrocoDuck:

I haven't tried your other suggestions yet but was able to give the avlinux livecd a go this afternoon. Boy, that livecd is *stuffed* with the goods! I am going to use that to try out a number of the audio apps that are already on my list of things to play with.

What struck me while running it was that it defaults jack settings to this:
Frames/period: 512
Sample rate: 44,100
Periods/buffer: 4
Latency: 46 ms

Of course there were no xruns with this setting.

Unfortunately, when I tried reducing the latency with what I have on my Arch system currently (256, 48000, 3) I started getting xruns at a rate of 1 every two minutes which was worse than on my Arch system where it was about 1 every five minutes. That was really disappointing as I'm given to conclude that the livecd had no better settings than what I already have, at least by default.

It also got me to think: should I be settling for latency in the region of 46 ms as the livecd had defaulted to? From all my readings I understood that I should be shooting for <10 ms, but given the hurdles I'm facing is, say, 30 to 40 ms, OK? I'd be glad to hear your views before I start a new round of experiments!

Thanks

There is no rule on latency actually and it depends on what you have to do. If you need to use sequencers only (without recording midi from controllers, only piano roll editing or similar) latency is not an issue and can be arbitrary large. If you want to record instruments and you can supply a monitor directly from the sound-card preamps latency is not an issue as well (this is what is usually done in recording studios, so that stability is not an issue). If you plan to use the computer for real time signal processing or midi/audio recording with monitor given by the sound-card outputs then latency must be low. If you reach less than 2 ms round-trip latency you are the king of low-latency. I think that less than 7 ms is good enough though. Use jack_iodelay to measure it as the latency given by qjackctl is the maths derived latency, calculated using the sampling - buffer variables, and does not includes latencies due to hardware components (converters in the sound-card, controllers, etc...).

An xrun every 5 mins is not that bad. I think you can improve, but a good start. I would advise to carefully choose the software as well. Like, qtractor offers good functionalities and it is way lighter than Ardour. When I use it I can hardly have xruns, but they pop quite often with Ardour 4.

Thanks CrocoDuck for your answer. For the time being I'm just using Guitarix to play my guitar with a number of IRs that I'm trying out - so not even at the stage of recording an instrument. Perhaps I'm fretting over something that isn't quite relevant to me yet.

Thanks also for all the links - I took a look at your blog and and other posts and will certainly be returning to take it all in.

Keep in mind that most xruns are inaudible. I have an old computer, and there are times
it is 'maxed out' and I can hear popping, while recording, but most often,
the recording itself doesn't have them.

And in the extreme moments when there is a an actual break in the recording,
if it's a track you want to keep, it's easy to redraw the resulting gap, using audacity.
Combining the slow playback, and zoom-in features of audacity, it's easy
to find and repair such things.

While you're editing, you can sometimes find some spikes in the audio,
that can be selected, and de-amplified or re-drawn much lower, allowing
the volume of the entire track to be amplified into the best range for
general listening. If a spike is only on one half of a stereo track,
audacity let's you split the stereo track, to edit the relevant half, and rejoin
the stereo halves later.

Not to mention boosting/lowering/removing moments of ham-fisted musicianship.
Which sadly, I am quite experienced at producing.

Thanks glowrak guy, I'll keep that in mind.

Not to mention boosting/lowering/removing moments of ham-fisted musicianship.
Which sadly, I am quite experienced at producing.

Ha ha, I like the way you say that! And more likely than not, I'm going to be a distinguished member of your tribe!

I also cheat... ehm... edit a lot, as you can tell from my tracks. However I always discard the tracks with xruns for a few reasons:

1) If I hear the pop and/or the associated audio interruption I am very distracted by it. Since I struggle to forge a groove I loose both time signature and intention and even if I can finish to record the part it never conforms to the groove I had in mind. I would have to separate parts of the take and move them around... which is much more time consuming than trying to get a new take that works.

By the way, this illustrate how critical stability is for me, as it always requires practice to get a part right. So, when it happens that you are recording it just as you have it in mind, you don't really want it to be ruined by artefacts that do not depend from you.

2) I left sometimes xruns around that were marked by Ardour by couldn't really hear them in the mix... just to find that playing the piece through different loudspeakers/headphones/systems gives them out. I then discarded the mixdown and done it over as I couldn't stand it.

Reason 1 is a consequence of the playing style I have and might be subjective, especially if you play less groove oriented music genres (for example, I didn't worry about groove that much when I recorded my sh**ty metal track). Reason 2 is a little more objective, but you can fix the problem as suggested by glowark guy.

However,

3) I prefer to focus on the musical side only of playing and editing. That is, my setup is intended to allow me to work without introducing artefacts, so that I can concentrate on the music and not on fixing problems that shouldn't be there in the first place.

So, my humble advice is to try to get the most out of stability if you need to record.

If you just need amp simulation and effects for fun / live performances, if the xruns are not too distracting or inaudible, you can go with them pretty well.

Thanks for your thoughts, Crocoduck - much appreciated.

Though I'm a total newbie, the xruns I'm getting do distract me (from the noise that I'm making ) Also, despite my newbie status, I am noticing (or maybe I imagine I am) the difference between lower and higher latencies. So it looks like I will have to continue my search for reduced xruns , and clearly also continue to read up more on the various concepts and applications involved. There are still too many things that I don't understand very well, which isn't helped much by my very severely limited guitar playing abilities!

The one thing I really enjoy is playing with a number of IRs that I have downloaded. I have so far the God's Cabinet, Goddamn and 7 Deadly Sins collections, plus more covering Peavey Vypyr, Krankenstein, Randall and the Mesa Boogie Mk V. They work so well with Guitarix!

More than anything else I've set my focus on getting my guitar chops together. So far I'm getting to practice about 3 hours plus a day and I'm feeling really good about it, even though musicality is still some ways off!

Following my test and similar results with the AVLinux livecd, I was concluding that my configuration wasn't the problem, and that perhaps it was some BIOS setting. I nearly left it at that, but (being the tireless sort) I thought I should give another distro a shot, so I settled on KXStudio.

And well, on the KXStudio livecd I'm getting latencies of 2 ms with nary an xrun in sight. The only funny thing is that when I increase frames/period from 128 to 256, or periods/buffer from 2 to 4 or 4 to 8, the latency doesn't seem to increase by much, as it tends to do on my Arch system.

The main differences that I've noticed between KXStudio and my setup are:
Kernel: Ubuntu low latency kernel vs my standard linux kernel
Cadence vs qjackctl on my system
Inclusion of snd-aloop jack bridge vs no bridges on mine.

Let's see - as I investigate some more.

I've only read through this thread briefly, and didn't see one suggestion (although you may have tried it already). I used to use an Alesis USB mixer that was very fussy as regards xruns with jack1 and with jack2 in its default asynchronous mode. Using jack2 in Synchronous mode made it work beautifully. Try jack2 started via "/usr/bin/jackd -S" and see if that helps.

Thanks @eric71. I may have tried that already, but will remember to try it again when I get back to jack2.