Trying to make a good 'narrator' voice sound

[kozikowski]

<<<How permissive should I be of an occasional red?>>>

Occasional red being a VU meter red or a digital signal meter hitting "0?" US-ANSI VU meters are supposed to gently hit red on occasion. That's how they work. Digital red represents permanent destruction of the quality of the sound. There is no tolerance and no recovery after it happens in the Exported show.

This is why you produce the work low and then at the end, when you get it perfect, set the delivered show to the desired peak values with Amplify -- not Normalize. You know "Normalize" messes up left-to-right relationships, right? Amplify doesn't. Normalize is different in many other sound programs. It's one of Audacity's nasty surprises.

<<<Then I can take advantage of proximity effect variation, when I'm pasting together the final.>>>

Yes, but the other thing that does is give you the announcer three inches from your ear effect -- or announcing into a telephone. Neither sounds particularly good. Many bad things happen when you're too close to the mic -- even non-directional mics which don't have proximity effect.

That's not to say you can't do that for special effects. I, in my deep voice, once did a passable woman by getting really close to the mic and deliver the lines in a breathy whisper.

<<<I think there's pretty tight limits what I can do with the EQ or the "bass boost". It seems that taking bass out is easy, but when putting it in, you get muddy fuzz pretty quick.>>>

Are you going into overload? See: produce the show low and set final level just before delivery.

You can only boost what's already there. The system can't make it up on the fly -- although there are software packages that people keep claiming can do that. Does the work sound like the actor? I'm guessing yes, and do keep in mind you're using a microphone designed for live performance screaming, not studio capture. The first time you use a higher quality microphone you will be surprised at the work you don't have to do to make the capture presentable.

One further note, Chris does have problems with extreme beginnings and ends of the performance. It's common to add about a minute or so of silence to both ends so the software has someplace to go to catch its breath without affecting the show.

Koz

[kozikowski]

Wait. It's not a minute. I think it's ten seconds. I gotta go look that up.

Koz

[breslin]

You know "Normalize" messes up left-to-right relationships, right?

Oh dear oh dear. I don't even know what a "left-to-right relationship" is. I really need to get some proper training in this field! Well, I'll pick up as fast as I can.

[kozikowski]

It isn't that crazy. Amplify rips through the whole show and adjusts one master volume based on the one loudest sound from either side.

Normalize adjusts the volume of left and right individually. The upshot of that can be a startling distortion of, for example, where the instruments are in your head while you're wearing headphones. If you went to great effort to cause the trumpets on the right to be more prominent, Normalize may try to "fix" that for you and even them out.

Koz

[breslin]

Ah, that makes very good sense, thank you. Like a typical beginner, I will imagine mysteries in simple things. For example, I was leery of raising amplification towards the clip level, lowering it again, raising it again, lowering again... I imagined that some frequencies would be clipped out anyhow, and flipping up and down would incrementally decrease the sound quality. I now think that this might be true for analog recording, but the digital source is unaffected by this (unless you *do* go past the clip-point).

That said, I don't understand why we go for -1.0 db amplification, rather than 0.0 amplification (which should be safe, according to the mathematics).

[kozikowski]

I got a minute to address the bass problem.

http://www.kozco.com/tech/audacity/Voic ... sBoost.wav

I only did the first four phrases. Note the level doesn't change.

Koz

[kozikowski]

<<<I now think that this might be true for analog recording, >>>

Analog recordings are constantly crammed between tape saturation at the top end and oxide and electronic noise at the lower. In Digital, the overload point is always right there next to you, but there effectively isn't a noise floor. 16-bit noise floor is -96dB. The absolute quiet limit of human hearing is about -60dB and the electronic limit is 36dB below that.

The broadcast overall average program level is -20dB (in the US) leaving a very generous amount of room for loudness increase with no distortion and still maintaining 70dB noise floor. Tape recordists would have killed for specifications like that. It is magnitudes better than you can do with the best low-noise tape on a perfectly adjusted machine.

So produce your brains out, but do it generously lower than overload, then Amplify as the last step to get the deliverable product. If you're constantly jacking the volume up and down and smacking zero, you're definitely doing it wrong.

<<<That said, I don't understand why we go for -1.0 db amplification, rather than 0.0 amplification (which should be safe, according to the mathematics).>>>

In digital, yes. But sooner or later it gets converted to analog and that step is where the process can become unstable.

Koz

[kozikowski]

<<<Note the level doesn't change.>>>

We also note that listening on my laptop speakers, there is no difference between the first four phrases and the rest of the clip. On my killer sound system® there is a substantial change.

Koz

[breslin]

Very interesting treatment of the bass issue!

I think you amplified with a level of -7 (to give you room to work with), then applied the eq. I tried to imitate it (still working on my ear!), and ended up with something that looked like this:



It still sounds like I've made some error in the eq, but with my untrained ears and my wimpy speakers, I'm having trouble locating the error. I need to get new speakers, and train my ears up to learn which frequencies are which. I guess the latter is just a lot of practice on the eq.

On the subject of speakers, should I be looking towards high-quality computer speakers, or (I think more likely) a reasonable power amp and a set of studio monitors?

[kozikowski]

Yes, you really can't hear what you're doing unless the sound system is up to it. There was a commercial a while back of a guy cutting together his movie on his laptop on the flight back from the live shoot. I guarantee he didn't cut the sound like that.

That's my system...

http://www.kozco.com/mytv/mytv.html

The sound is described in the text. Nice headphones are sometimes indicated.

Somewhere on your machine is a file called EQCurves.xml It likes to hide behind fancy-pants labels, but it's just a text file and it will open up in TextEdit or NotePad. Paste this code into that file and save it. Make sure your editor is not set for Rich Text. Only Plain Text.

The file has many other programmed curves and it should be no problem to follow the rhythm and style of the other filters. Spaces and carriage returns do not matter in XML code, so they're used for neatness and maintainability.

And compulsiveness. I can't sleep until each '"f" lines up.

<curve name="Endoria">
<point f="21.419463020801" d="7.837839126587"/>
<point f="30.178767256116" d="8.648649215698"/>
<point f="40.620196833280" d="9.054054260254"/>
<point f="64.160230052647" d="9.459459304810"/>
<point f="107.301375083257" d="8.783782958984"/>
<point f="142.785043486344" d="5.405405044556"/>
<point f="171.432150744924" d="1.756757736206"/>
<point f="217.930148818174" d="-1.351350784302"/>
<point f="255.741186981536" d="-1.891891479492"/>
</curve>

And it should look something like this...

http://www.kozco.com/tech/audacity/endoria.jpg

The negative offset is how I got the overall volume reduction. There's a lot more design in there than there appears. There are no sharp bends and the boost curve was created using the droop curve of your microphone as a model and starting point.

Koz