Audacity Forum

I'm doing a lot of homegrown voiceover these days, and as wonderful as Audacity is, I can't figure out how to take my excellent recording and process it so it sounds truly professional.

I'm running an AT4040 Cardiod Condenser mic through a Lamda Lexicon interface unit. I'm recording in my room, which is fairly noisy for an audio booth. Once I use Noise Removal and Normalize the track to get it nice and loud, I get a nice, clean recording that sounds just like me, but is several steps of polish away from sounding truly pro.

I've tried running an Equalization pass with the very lowest and highest sounds dropped out, and that seems to help, but I was wondering if anyone out there had any ideas or tips on how to make the recording go to the next level.

I've include a sample of my voice with only Normalization and Noise Removal applied, and would love to get some feedback.

(I do have to say that I have to Normalize the audio to get it up to a usable volume. I wonder if I'm recording at too low a volume. I have the Recording input slider all the way up in Audacity, and my Lexicon Mic1 level is up at about 3/4. Any higher and it starts to peak. The 'low cut' switch on my mic is on the '---' setting (as opposed to the '/--' setting) and the 'Pad' is at 0db (not -10db). So I think I'm recording at a high enough level, but I'm not sure because I always have to turn it up thru Normalizing or using the track volume db adjuster to get it to a usable level.)

DoomMunky wrote:I'm recording in my room, which is fairly noisy for an audio booth.

Anything you can do about that?
If you have to use noise reduction, you're putting yourself one notch behind before you've done anything else.

What is your background noise level metering at, and what is your peak recording level (before Normalising, or anything else)?

DoomMunky wrote:The 'low cut' switch on my mic is on the '---' setting (as opposed to the '/--' setting)

For voice, I would set it to /--
This will apply a gentle low frequency roll off, below the frequencies of your voice.
It should not affect the sound of your voice, but will help avoid background rumbles and other unwanted sub-bass.

What is your recording distance (microphone to mouth)?
Do you use a pop shield?

I also use the AT4040. I've been experimenting with distance and angle. Some recommend that you record slightly to the side with the AT4040, though I like the sound with it directly in line with my mouth.

Listening to the mp3 you provided, I could hear quite a bit of the room and not so much of your voice; the sample doesn't have the "tight" sound that is desired for voice-over work (if reverb is desired, it can be added after the fact). If you're not overly concerned with aesthetics, I recommend you purchase moving blankets from a place such as Lowe's and hang them on the walls. That's what I did, and the difference was night and day.

Thanks for the responses, guys!

I AM using a pop-shield, and I'm keeping my mouth between 6-10inches from my mic. I'll experiment more with talking to the side of it, too. My background noise barely shows up, but I'll see if I can get some sort of db reading on it.

And I think I'm going to move my mic setup into my closet, put up some blankets (or cleverly hang clothes) and see what it's like. I'll do that soon and have some more samples for you soon.

Thanks for the ideas and feedback. It's hard to do this alone, and hearing others' ideas and suggestions (and similar problems) is reassuring and helpful.

Something I forgot to mention is that the AT4040 picks up a lot of room noise anyway, so you may need to move in a touch closer to get less of the room.

How are you listening to your recordings? I just bought the Audio-Technica ATH-M50 headphones, and they are good at revealing details you wouldn't hear otherwise, especially room noise (that's how I noticed it with your sample).

Interesting that the mic is 'known' for certain behavior. I wasn't aware of that. Where do you get that kind of info? I'd love to see a 'best practices' list for this particular mic...

I listen to the final Audacity product through some nice Creative earbuds. I may invest in some studio headphones down the road. I don't monitor my input. I prefer to just record for a minute or so, do a bunch of different takes without worrying too much about how it sounds, then listen and adjust as necessary, occasionally re-recording.

From what you're saying, it sounds like I'd be well served by two main things: 1) Quieting my recording environment as much as possible (duh) and 2) getting in closer to my mic (and turning down my input gain a little to compensate).

getting in closer to my mic (and turning down my input gain a little to compensate)

On input gain, read this. To sum it up, with 24-bit recording, you have more headroom on the low end and don't need to be worried as much about recording too softly. It's easy to normalize the track to whatever level you want.

Interesting! Does that mean it's easy to fix clipping/distortion/peaking if I go too hot? I don't know how to do that...

My main problem is that I peak out and distort when I get louder, which happens semi-regularly in voiceover recording. Some lines are very dynamic, and have quiet parts interspersed with louder things within 5 seconds of each other.

Where is a good level to set the input at? Should I be adjusting it for every single different recording I make, or is it better to set a limit where it peaks only when I'm almost shouting, and then just compensate for low volume with Normalization and so on? (I am pretty much just making it up as I go along, so the more basic, general recording advice you can give the better; stuff like how to set input levels. I'm ONLY doing voice recording, so that makes it easier.)

DoomMunky wrote:Interesting! Does that mean it's easy to fix clipping/distortion/peaking if I go too hot?

No, quite the opposite.
It means that when recording with a high bit depth (24 or 32 bit) you can afford to leave lots of headroom without sacrificing any dynamic range.
You should aim to have your analogue hardware (including the analogue inputs of your sound card) working within the range they are designed for, but when you hit the digital domain, the dynamic range of 32 bit float is so huge that if your highest peak is at -12 dB, there are still more than enough bits below that to accurately reproduce every detail.

There are really 2 parts to this subject - the design sensitivity of the analogue equipment, and then the digital realm.

In the old days of audio tape, it was necessary to try and squeeze every last dB possible onto the tape by running as close to the red as possible. That was because of the limited dynamic range of audio tape. By recording as loud as possible, the noise floor of the tape (background hiss), would be relatively quieter.
Today, with 32 bit digital recording, Even if we "waste" the loudest 12 dB possible, the digital noise floor is still way below the noise floor of any of the analogue equipment that is being recorded.

However, that does not mean that you can just record everything quietly and boost it up with Normalization and expect a top quality recording. Microphones, pre-amplifiers, mixing desks and sound card inputs all have there own noise floor. Each link in the audio chain should ideally be running within its design parameters.

Some years ago, I wanted to record the ticking of a wrist watch. At that time, the best microphone that I had was a Sure SM58 dynamic vocal microphone and a cheap realistic mixer. Even if I recorded onto a state-of-the-art digital recorder, my results would have been poor, simply because the self noise of the microphone and mixing desk were almost as loud as the ticking watch. What I would have needed would be a very sensitive, low noise microphone, and a high gain, low noise pre-amplifier.

Similarly, if you try to record a bass drum from a rock drum kit with a highly sensitive condenser microphone, it is very likely that irrespective of what volume you set the microphone pre-amplifier at, the recording will still come out sounding distorted because the microphone is being overdriven.

When recording, I try to use the right microphone for the job, and assuming that I have it plugged directly into a mixing desk, I will adjust the input gain so that the pre-fade (input) level is peaking close to the optimum level (my mixing desk will go happily up to +12dB before clipping, so I can drive the input close to 0dB with most material. I will also adjust the output from the mixing desk so that it suits the input of the recorder, which in the case of my sound card is to peak no higher than -3dB as an absolute maximum. Finally I set the recording levels for the digital audio recorder (Audacity), and here I can finally relax a bit - with a recording level at around -16 dB, I have lots of headroom, and the (very low) self noise of the analogue equipment is still faithfully reproduced.

Excellent breakdown, thank you!

For someone like me, with my AT4040 and my lonely Lexicon Lambda interface device (no fancy mixers here!) I only have two input settings to worry about. The mic-in to the Lamda, and the recording level of Audacity, which I believe receives the direct input of the Lambda (no separate output setting).

What I'll try doing is setting the Lambda Mic-in so it gets reasonably close to peaking during normal, energetic speaking, then setting the Audacity recording level to peak at about -12db. I'll see what that's like, then Normalize as needed.