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USB vs. latency vs. using my PC as an amp...

meatwad

meatwad

Apr 9, 2008
2,734
1,033
Smallville, USA
#1
Here goes a total noob question, I'm sure. The more I try to read and find the answer to my question, the more confused I get, so here it goes -

I want to use my PC as an amp for both my bass and geetar! I have Guitar Rig 2 (along with other ancient programs like Cool Edit Pro 2, Fruity Loops, ect..), and want to be able to plug my instrument into an interface, which plugs into my PC's usb port, and monitor through my PC's speakers without latency.

I have recorded in the past by splitting my bass/guitar's signal, and simply plugging one side into the PC's 1/8" microphone jack for recording, and running the other side to an amp for monitoring.

Will any of the interfaces on the market do this? Maybe suggest me something that will? I'm not married to the software I already have, and am willing to try something new if it will make this happen for me.

Thanks! Sorry if this is a dead horse topic.
Wade
 
#2
I can't think of an interface that wouldn't do exactly this. I never have any noticeable latency when I record with my Firewire interface and I can't imagine USB being that much different for a single instrument.

Whatever software you use make sure you set the input to mono instead of stereo so that the signal will come through both the left and right speakers.
 
#3
I believe there is an inherit difference in USB vs Firewire latency. I remember reading someplace that the way the data protocols work USB (particularly 1.1) has that issue. Although my experience it is not noticeable. I have both USB and Firewire interfaces and for the most part in Linux can dial them into pretty much real time latency levels.

Since you are saying PC I am assuming you mean Windows. First thing first, is ASIO installed. It needs to be if you want real time latency levels. Next is it Windows 7. I have seen and heard issues with Windows 7 and latency. Step one just turn off all the extra windows decoration and animation stuff. That will help a lot, but ASIO will still be needed.

I have used a computer for effects. Now granted I use linux. Linux when done right can have some of the best latency numbers around.

Ken
 
#6
meatwad said:
Interesting stuff. Here's some specs for my PC (laptop) -

Acer Aspire 5532
Windows 7
AMD Athlon 64 TF-20 1.6 GHz processor
3GB RAM
two USB but no Firewire inputs

I keep it running clean, more towards speed than pretty.
Click to expand...

You are asking quite a bit from a single core 1.6 ghz processor. I have gotten it to work on 1.6 ghz Atom but that was tweaking the Linux install and it still couldn't handle heavier weight effects that required note tracking like an octaver or harmonizer. It would cause Jack to have xruns everywhere (for the non geek that means the low latency sound server was erroring out and causing a nasty static stuttering sound.

In the Windows world I don't think that will cut it. The TF-20 is a low end processor. Quite frankly single core processors should be off the shopping list even for budget computers in this day and age.

Ken
 
#7
meatwad said:
This would be a very important first thing. Thanks devilock!

:bassist:
Click to expand...

It really is, as far as Windows goes. And to add more info for the less than geek. Windows does not come with a real time kernel (kernel is the part of the operating system that handles instructions to the CPU). What happens is when an interrupt happens (IRQ) such as audio or midi events the program sends a request to the kernel to process whatever it needs to do with that instruction, like send the audio back. Without a real time kernel that audio request can be shoved behind many other requests to the kernel (and a lot are happening even at idle). ASIO is a means to make the Windows kernel behave in realtime for audio instructions. Not being a big Windows guy I don't know the tech.

Mac's to my knowledge do actually come with a real time kernel for desktop versions of OS X. I don't think this is the case for OS X Server, at least I hope that is not the case. That being said Mac's are real time for audio out of the box which is one of the reasons they have proliferated so well in the pro studio environment.

Linux typically does not have a real time kernel. However numerous projects and distros exist that changes this making the kernel real time. What this means is that a person with the know how can make Linux easily match and even beat a Mac as far as audio latency figures. And linux will do that at a lower cost (typically free). Linux does not come boxed usually with a real time kernel as it is used often in server environments and a real time kernel is not desireable for a server (note my OS X server comment above). However as far as realtime on the desktop no problem. However this default kernel choice is why at present IOS (iphone and ipad) destroys android in the music app arena. Google has not seen fit to use a real time kernel for android and that is a huge mistake considering the target platforms.

Other Unix's (Solaris, Open Indiana, the BSD's, etc) are even more geared towards the server world and typically don't ever need a realtime kernel. You can get BSD to run almost realtime with the factory kernel and Jack in most cases. Otherwise you will have to re-compile the kernel manually to enable that. If previous tech comments in this post made your head spin then compiling your own customized BSD kernel is outside your ken.

As a point I am typing from a laptop I use for mobile recording. It is a Toshiba with 2 gigs of ram and a 1.8 ghz dual core. I use a Presonus Firepod firewire interface. Linux install is Fedora 14 with the Planet CCRMA repos installed. I can get near 0 latency easily. However in the future I plan to move towards Arch linux as my choice for Linux recording OS.

Ken
 
#8
devilock76 said:
You are asking quite a bit from a single core 1.6 ghz processor. I have gotten it to work on 1.6 ghz Atom but that was tweaking the Linux install and it still couldn't handle heavier weight effects that required note tracking like an octaver or harmonizer. It would cause Jack to have xruns everywhere (for the non geek that means the low latency sound server was erroring out and causing a nasty static stuttering sound.

In the Windows world I don't think that will cut it. The TF-20 is a low end processor. Quite frankly single core processors should be off the shopping list even for budget computers in this day and age.

Ken
Click to expand...

True, this laptop was on the top of list of Best Buy's website, when sorted "priced lowest to highest", 1 1/2 years ago. I have heavily considered lately building a desktop PC dedicated to home recording, and LINUX definitely sounds like a great idea for the OS. I have over a Terabyte of harddrive space on the shelf already, so sounds like I should just consider that as a start towards achieving my goal...

Or I may buy a Mac. Who knows.
 
#9
meatwad said:
True, this laptop was on the top of list of Best Buy's website, when sorted "priced lowest to highest", 1 1/2 years ago. I have heavily considered lately building a desktop PC dedicated to home recording, and LINUX definitely sounds like a great idea for the OS. I have over a Terabyte of harddrive space on the shelf already, so sounds like I should just consider that as a start towards achieving my goal...

Or I may buy a Mac. Who knows.
Click to expand...

If you are thinking about using Linux for audio needs then you should roll on over to Invalid Link Removed. There is a wealth of much smarter people than me over there that I learned a lot from. I have to work on PC's for the day job, I have them in the house because our printer of choice just will not cooperate with Linux.

For me Windows is here only for the rarely needed device compatibility and the fact that since I have to deal with Windows for my career it is better to have at least one Windows machine in the house. Otherwise pretty much everything else here is Linux or BSD.

One day I might go mac here. I really like them and it is something to consider. I view Linux vs Mac like this. I can make Linux every bit as good as a mac for little to no monetary cost, the cost is my time. At some point I may not like spending that time so I will spend cash and get a mac. It is the choice of time vs money for me as I already have the skill set and experience to deal with Linux for whatever I need, from audio recording mobile or in house to running a development server out of my bonus room.

The one downside to mac is cost, but with that cost I get more software and hardware products from major manufacturers for professional audio that is a cost as well. But then again take the firepod I used. I had to research to find that and that it could be supported under Linux.

The point is if you go the linux route there will be some study and tweak time, but if you are willing to do it you will get mac level and better audio performance for much less monetary cost. If you go mac your purchase to recording time window will be faster. However I say this and have first hand witnessed studios have major malfunctions in their mac gear that if it was Linux I could have dealt with simply if it even ever happened in the first place.

In the end you might want to check out some linux stations and their music software and see if it even jives with you. In the end you are doing it to make music so leaving all biases aside, whatever helps you work on music better is the choice to make, not the recommendation of countless geek fanboys, myself included.

Ken
 
#10
Your plug-ins will also have a certain amount of latency built in too. It's just the way it is. The program uses the resources aloted to it, to process a live incoming signal.

10ms of latency you and everybody else will notice. Under 4ms and you should be okay.

I've been doing this stuff for so long that I can hear 4ms. But I dont let it bother me.

The best advice I can give you is get a better laptop, even used. A dual core Turion or C2D is essential. 4 gigs ddr2 and if possible a 7200 rpm hard drive, they'll be a little more expensive than the usual 5400's.

And as for 1394 vs. usb. 1394 all the way. Heck, MIDI is faster than USB. USB is around 1 to 2 midi messages slow. Even USB 2. I don't know yet about USB 3 and its inherent latency.

I was about to mention a pcmcia or pcm express, but your lappy doesn't have those.
 
#11
Thanks for all the fast responses, guys! I think the LINUX idea sounds much more doable, although it will increase the learning curve by quite a bit. I've really lost touch with building computers and hardware specs since somewhere between the demise of AGP video cards and the beginning of Vista, so I have much catching up to do.

I guess it's time to dig in. Thanks again!
 
#12
meatwad said:
Thanks for all the fast responses, guys! I think the LINUX idea sounds much more doable, although it will increase the learning curve by quite a bit. I've really lost touch with building computers and hardware specs since somewhere between the demise of AGP video cards and the beginning of Vista, so I have much catching up to do.

I guess it's time to dig in. Thanks again!
Click to expand...

Ok well if you want to try linux audio then first stop is ubuntu studio. It isn't what I would use for my own machines but is one of the best ways to get your feet wet in linux audio.

Ken
 
#13
Not to dis-suade you from trying linux, because it is great and everything said here is true. BUT. You should be able to do what you are wanting with nearly any off the shelf dual core or better PC and firewire interface. I'm running a 5 year old HP desktop with windows XP, and get I less than 4 ms latency running guitar rig 4 on my PC. Using presonus firebox. works like a champ.

Just another opinion for you to chew on.
 
#14
MrTAteMyBalls said:
Not to dis-suade you from trying linux, because it is great and everything said here is true. BUT. You should be able to do what you are wanting with nearly any off the shelf dual core or better PC and firewire interface. I'm running a 5 year old HP desktop with windows XP, and get I less than 4 ms latency running guitar rig 4 on my PC. Using presonus firebox. works like a champ.

Just another opinion for you to chew on.
Click to expand...

Correct, although with Linux I have been able to get USB to behave at extreme low latency. I think the big key here is XP and sadly XP is still the most recording friendly Windows version out there. But yes a typical dual core with ASIO installed will get you a lot closer. And to be clear, even though I think with the computer you mentioned Linux might get you closer, it is still asking a lot as the bottle neck in your setup is still that single core processor, regardless of what OS you use.

Ken
 
#15
A to D to D to A

It is impossible to eliminate latency when monitoring through software. Impossible.

It is possible to reduce it levels that most people find acceptable, but this requires a powerful enough machine, a good interface, and software that does its job adequately. At the end of the day, isn't easier to just monitor in the analog domain?
 
#16
silky smoove said:
A to D to D to A

It is impossible to eliminate latency when monitoring through software. Impossible.

It is possible to reduce it levels that most people find acceptable, but this requires a powerful enough machine, a good interface, and software that does its job adequately. At the end of the day, isn't easier to just monitor in the analog domain?
Click to expand...

The key is perceptible latency. And to muddle the issue this is exactly what all those digital mfx units do every day.

Ken
 
#17
You can get latency down to workable levels with a half decent pc (yours is on the edge but should still be doable) and an interface made in the last 5 or so years (further, to be honest, but I'm just covering my back :) ).

Firewire/USB won't make a difference unless you are recording lots of tracks, the trouble with USB comes with large track counts and lots of info going back and fourth to interfaces etc. Simple setups with 1 track monitoring at a time, you're ok, and USB will be able to hit acceptable latency.

you don't need to leave windows to get very acceptable latency...

Plugins do cause latency, many an almost imperceptable amount (10 or so samples, which is about a quarter of a millisecond at the slowest normal sample rate). Plugs such as limiters, anything with look ahead function, convolution, modern fancy reverbs and anything which sucks a lot of cpu normally has a larger latency. Enabling plug in delay compensation (most of the major DAW's have this) will sort this out for playback, but will cause more problems when tracking. Basically, a latency compensation engine looks at the slowest channel strip (after all plugins are considered) and works out how slow it actually is, then delays all other tracks so they are in line with this one.

For that reason its good to use only low latency plugs when tracking, or have 'plug in delay compensation' turned OFF!

In reality, using eq's and compressors and reverbs on a bus will not effect you at all, but its good practice to have it turned off anyway while tracking.

ASIO is good, but all interfaces now come with a driver that expands upon what asio does, and will be able to acheive better results with a driver that is tailored more specifically to the soundcard in question.

There is a lot of scaremongering offered about regarding latency. I had a dell p4 2.0 Ghz working at great latency (about 4ms) 10 years ago without any troubles, hell at the time I didn't have a clue of any of the technical points I just had a well kept pc and set the buffer size low and kept crazy plug-ins to a minimum!

These days, if you can't get it to work down low then theres something wrong with your machine or its not set up correctly!

I've installed many studios, worked in big facilities and made more records than I can remember and the sort of workarounds people are suggesting in this thread, while well meant and based on truth, seem a bit ludocris to me!

Granted, for a proper studio you will want either analog, pro tools HD or other direct monitoring platform as the basis but a digital monitoring solution is good enough for a lot of small sized studios I've worked in, most of them running old computers :) (upgrading a computer is the last thing on a studio owners mind, too much fancy stuff to buy instead!)

PLEASE do not go into the process of changing OS or buying loads more gear before you have got a cheap soundcard and set the buffer size and DAW pref's correctly and evaluated whether the latency is low enough to work with.
 
#19
devilock76 said:
No matter what your computer did 10 years ago don't forget that all the software makers are gearing their software for machines with larger specs than what you had 10 years ago.

Ken
Click to expand...

Hmm, not really, not when it comes to the DAW itself. The imprint the DAW has on a system has always been pretty small, and these days its tiny in comparison to available power! The DAW is just a shell for processes you set up of course. Its these processes that eat up the power. Logic of today requires not that much more power than Logic 5, for example.

Even many plugins, eq's and the like, still don't tax the computer that much, with the exception of newer technology plugins like convolution plugs, linear phase plugs and the like as well as a few of the more fancy reverbs and eq's/comps effects.

So as long as you are keeping plugins under control the main strain on your system is going to be ONLY getting the audio in and out at the selected buffer size (and as the project builds up getting audio to and from the hardrive as well), and the power required to do that has not changed that much at all, it even requires a bit less these days with modern soundcards/drivers.

Case in point I had to dig out my old pc this year when I had my laptop stolen, I put a new copy of pro tools mpowered on it and it ran better than the version of cubase sx 1 I had bought 10 years ago! It still afforded me the same I/O track count at the same latency with no bother even on a much newer DAW and using plugs released last year.

Anyway, don't mean to be argumentative, its just this guy has asked a relatively basic question and has received a lot of suggestions pitched at a much higher level than he needs. Its good to understand this stuff and if you are making a really nice little studio then direct monitoring is something to be seriously looked at for a number of reasons, but for most users just knowing how to set the buffer size and to keep plugins and external computer processes down is enough to easily get 3ms latency these days. That and a few computer housekeeping tips!
 

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