Micing the amp and latency issues

[KrisWarrington]

I usually do professional recordings but just recently I bought a condensor mic (usb) to try a whirl at doing my own recordings and using it to make you tube videos. I am also using Adobe Audition 3.0 (which has a latency compensator) I am running into several problems however and I honestly don't have the vast knowledge to fix it.

First off the recordings are never in sync so what i would do is lay down a scratch track and record the whole song based on that scratch track so even though it has a small about of lag the other tracks are in sync.

Second off there is too much background noise so what I do is cover my amp and mic with a blanket and use some effects from adobe audition (noise removal ect) but that just makes the recording sounds like **** because it cuts in and out and if you record too loud the whole recording just clips and is over powered.

Any suggestions would be nice. Here is a list of what equiptment I am using

MXL Pro-1B Microphone
Adobe Audition

Kustom 1200HD Head
DBX 166xs Noise Gate/Compressor/Limiter
DBX 215s Equalizer
John Moyer G810H Cab

[KrisWarrington]

No body?

[silky smoove]

Latency, in your situation is something you're going to have to live with. You've got a mic that goes in through USB. That means that an A/D conversion happens at the mic. That equates to latency. After the signal comes in, if you have any software monitoring going on, not to mention the audio from the existing tracks, it has to go through a D/A conversion to come out of your speakers. That's more latency.

Latency compensation is nice, and it's getting better all the time, but the only way to record in a latency-free environment is to keep your monitoring signal path analog. Given the type of mic you're using, I don't see how that's possible. You can minimize the problem with the latency compensator you're describing, and running as large of an input buffer as your system will allow. Check to see if you can record in a low latency mode. These typically disable plug-ins while you're in that mode, but if it's just for recording, you should be fine. Disable it after you've captured the recording and you'll have all your plug-ins back.

As for the noise removal, your recordings can only be as good as the environment in which they were recorded. An easy, and cheap solution is to build a small vocal booth. There are great plans online for building one using PVC pipe, pipe clamps, and heavy mover's blankets. They don't offer total isolation, but they will serve to remove noise from a computer's fan, assuming the computer isn't in the booth with you ,and will effectively deaden the space (i.e. echoes, standing waves, etc.). I built one as a project for under $100, and it only took about 2 hours.

[JimmyM]

Tru dat. It also helps to have not only an analog path but a computer with a fair amount of RAM, as music playback with a lot of tracks eats RAM.

[Charling]

That is true, but realistically you can get latency down to more than workable levels with a digital interface these days. I was getting 4ms back as far as 2000 with a modest pc and cheap soundcard, for most home users you should be able to get latency down to the point where it is VERY hard to notice.

For discerning studio applications direct monitoring with either a desk, pro tools hd or a good soundcard set up correctly is preferable, but many studios work with software monitoring without any complaints.

These days I can hit 2 ms in day to day use, it takes a musician with a pretty special amount of feel to be put off by that!

Try fiddling with the hardware buffer size, that acheives lower latency at the expense of more computer power, but these days even an off the shelf home pc has enough power to manage low latency with ease.

A latency compensator relates to input and output latency, not monitoring latency, and most software has this built in and turned on as standard. Sometimes it is not (or has been turned off) though, so if tracks are not lining up then look in preferences to make sure 'adjust for record latency' or similar is enabled! This will not effect the latency you get when recording in real time however, thats a different thing governed by the buffer size. Also turn off plug-in-delay compensation, as that will add an amount of latency to monitoring playback equal to the slowest plugin you have open.

As for noise, yeah, its down to the environment you record in mainly. If the noise is in the room , turn up the amp so it acheives a better ratio of desired to undesired sound, if its electrical hum and hiss that you cant hear in the room then play around with the mics gain, trying to get as much of the desired signal in while getting out of the 'noisy' range of the gear you are using, its all about gain-staging, ie making sure each volume knob along the way is set at a nice healthy level before distortion to elevate each stage out of the noise floor (ie the inherent background noise of a system)

O

[silky smoove]

While I agree that 2-4ms of latency is achievable, and not noticeable by the majority of musicians out there, I really don't see it being likely in this situation. The culprits being a USB mic, and a relatively old DAW that was fairly limited except for the built-in wave editor, which was quite extensive.

I think the OP can make the best of the gear he has, but my suspicion is that he may not ever be fully satisfied with the amount of system latency he has to deal with.

[duderasta]

I know in audition playing around with the buffer size can reduce latency, I don't recall the exact settings anymore, I use reaper now. A decent entry level interface with native ASIO drivers will help more than anything though.

[Mykk]

I know this isn't the ideal situation or the same software and componants as the OP.

What I do is play the original sound file via the computer (Media Player or otherwise) and set up a third party digital recorder not associated with the computer to record the bass cab. You play along with the original recording via monitors as the aux recorder is miced on your cab and than later move the bassline track from the aux recorder onto the computer and sync it up to the original recording.

This way works great for recording bass tracks to songs but only works if a cab is mic'd. Going line in direct and this method wouldn't provide an audible bass track while recording.

[Charling]

yeah, its far from ideal as far as latency goes, but very low levels SHOULD be achievable even given the gear the OP is using. The old DAW, for example, has no effect on latency, thats a function of the soundcard and computer power only!

A MODERN usb mic will be equipped with ADC technology of the last few years, and will be able to hit a near zero latency with no problems, its just a soundcard in a mic at the end of the day, and every soundcard for the last 10 years has been able to hit the values generally touted as 'near zero latency'. If my old £40 M-audio of 1999 can do it i'm sure a modern piece of gear has it nailed

That said, if its an OLD usb mic I'd hold less hope of it working out, if it comes from a time when near-zero latency technology was in its infancy for example.

Mykk, if the setup you have works for you then thats the main thing, but its a very roundabout way of achieving results easy to find on any modern single cheap interface setup on a pc. Have you looked properly at the buffer size settings on your soundcard? You should be able to get them to useable settings without the 3rd party device, if not then at least you can use your 3rd party device as a low latency monitoring solution that passes the audio on to the DAW in real time so theres no need for manual aligning!

I've spent a hell of a lot of hours of my life editing tracks for big releases, and theres nothing like uncertainty of where the track should lie originally to make you question where the groove is! If what you record is how you played along to the track in the first place you will get closer to a recording a proper groove instantly!