Hi everyone. I've been doing some very basic recording on my computer. I'm using free programs, Audacity and Kristal.

I've been having a very irritating problem lately. When recording, say, guitar, I run the guitar through my amp, and place a computer mic in front of the amp to record. One time I was working on a song. I recorded the melody, and then attempted to harmonize it... but the harmony was way off time. I figured it was a playing mistake, so I used the in-program metronome, and tried again... same thing happened.

This has only been a recent problem. I've done other recording with no such trouble, using the exact same equipment.

So is there some sort of delay that's causing this to happen? Any help you can provide is greatly appreciated.

Graeme

Not the mic's problem, but the latency in your DAW. Adjust the audio buffer size to get better latency. Actually a lot of times you'll just be able to adjust how much delay you want. The less delay the better for live recording, but then you risk dropouts. The more delay is better for mixing and listening back, but as you found out, real-time audio isn't so real-time then.

Not the mic's problem, but the latency in your DAW. Adjust the audio buffer size to get better latency. Actually a lot of times you'll just be able to adjust how much delay you want. The less delay the better for live recording, but then you risk dropouts. The more delay is better for mixing and listening back, but as you found out, real-time audio isn't so real-time then.
Thanks for the reply... but I didn't really understand any of it. :D

Is there any way the latency, DAW, audio buffer thing could be dumbed down?

Thanks,
Graeme :help:

I'm guessing Krystal has latency settings in some sort of 'preferences' menu. Use a smaller buffer size.

Specs of your machine? My guess is it can't handle audio recording. Plus, Audacity is a piece of crap. If you're on Windows, get N-Track. Oh, and at least a SM57 and a crappy Behringer mixer.

P.S. DAW = digital audio workstation, latency = the delay between playing a note and the computer's hardware interpreting the data and storing it to memory (lots of high-speed RAM helps this a LOT), and I don't really know how best to explain audio buffers.

P.S. DAW = digital audio workstation, latency = the delay between playing a note and the computer's hardware interpreting the data and storing it to memory (lots of high-speed RAM helps this a LOT), and I don't really know how best to explain audio buffers.

Well, latency is caused by a lot of various factors. The first thing to do is find the latency controls and find how low you can set them. RAM will help a little bit, but mainly in loading times and the overall bandwidth of your system. Otherwise, it's mainly the quality of your soundcard.

If you have a way of using zero-latency monitoring (providing you don't want realtime effects while you play), use that.

If you can adjust the actual audio buffer in KB or ms, put it to something low when recording, and higher when you're mixing.

Just remember that latency is something that we ALL have to deal with.

Thanks for all of the responses. As for my specs, I really have no idea. It is a rather new computer, and is damn fast, so I assume it's capable of recording and such.

Kristal does have a preferences menu, and there is an option for recoring buffers, which is set at ten out of twenty, and playback buffers, at six out of twenty.

So I assume the recording buffers must be lowered. Is how much simply a matter of preference?

Thanks again,
Graeme

It's not just a matter of how fast your computer is. Your soundcard has an awful lot to do with it. If you're using Soundblaster or Audigy, forget it. You should be using something specifically designed for recording. But STAY AWAY from M-Audio, their products are garbage.

Thanks for all of the responses. As for my specs, I really have no idea. It is a rather new computer, and is damn fast, so I assume it's capable of recording and such.

Kristal does have a preferences menu, and there is an option for recoring buffers, which is set at ten out of twenty, and playback buffers, at six out of twenty.

So I assume the recording buffers must be lowered. Is how much simply a matter of preference?

Thanks again,
Graeme

Set recording buffers to 2 or 3. Playback buffers could probably be 4.

Snarf, the Audigy (the old version), which I have used for a long time, is quite a decent soundcard, actually. Sure, no ASIO support, but ASIO is ridden with artifacts, and the latency isn't all that much better than what you can get from good latency settings (knowing what you're doing when you set your audio up). If you want ASIO and higher bitrate/samplerates for the Audigy you can always use the free KX drivers.

I'll admit some of M-Audio products are ones you'd want to stay away from. They are all plastic-y, but there's a few products that are nice. The M-Audio Ozone is a really nice portable unit, and that's not my opinion only, it's used widely in the pro music tech industry, I see it all the time.