Hello, folks. I was referred here from audacity-users. I'm trying to get started with Audacity and, to some extent, with audio editing. As this is the first GUI tool I've used, other than Goldwave (which I've only used slightly), I know enough about audio editing to shoot myself in the foot but not quite enough to look flashy while doing it. :) So what I've got, in addition to Audacity usability questions, are a few basic audio editing issues as well.
I've actually had quite a bit of trouble finding this list and its associated website, so I'd like to contribute any efforts to making it less difficult to discover, if possible. To that end, I'm trying to use Audacity to produce podcasts, and would like to do an accessibility-oriented tutorial for getting started with it, perhaps as a podcast so folks can actually hear how one goes about setting selections, contracting them, etc. Initially that was a bit confusing, though not so much once I got the hang of it.
But, anyway, onto the questions. :) It looks like there's quite an active community of plugin developers, or that there was at one time, and that there are several plugins for making inaccessible tasks less so. I've downloaded but not yet experimented with a collection whose name I don't recall for certain (Stereo Butterfly?) and it looks like there's a plugin to help with timeshifting. Thus far I've been inserting silence to timeshift segments. Are there any flaws with this approach? Will it make my projects larger as tracks contain various amounts of silence, or does Audacity simply note that there should be X amount of silence at certain points? Is there any way to use the timeshift tool accessibly, or is this plugin (whose name I can't specifically recall) the only way to go?
I'm combining recordings made on my mic with field recordings from an Edirol R-09. This unit adds a good amount of preamp hiss which I'm trying to figure out how to remove. Recording silence and passing that to the noise filter does a reasonable job, but the recording sounds much more digitally processed afterwards, so I'm thinking that EQ is the best way to go. I can't figure out how to access the EQ, however, as JAWS reports a lot of percentage slider bars, which I assume to be the bands, but I don't know to which frequency range each bar is associated. I saw a simple-looking EQ in the collection of plugins I snagged, but are there any accessible parametric EQ plugins that might help with this? How else might I go about removing this hiss in post-production? I'm experimenting with things on the recording side by disabling automatic gain control and setting a more conservative input level to minimize the processing added by the preamps.
Given a project where you have, say, one or two segments of speech recorded directly into the sound device, plus a number of other segments recorded in the field with slightly different levels, what is the best way of creating a recording with a uniform level that doesn't trash the ambience? I could normalize, but AFAIK this doesn't set an average level. I could compress slightly or use the leveler, but I'm not quite sure how these two differ. What I'd like is for sounds in my immediate environment to be clearly audible with background sounds adding flavor but not necessarily overwhelming my main focus. Is slight compression the key to achieving this? And what about the situation where you've got a number of different segments recorded in different environments and you don't want your listeners adjusting their speakers for every change of scenery? Right now I'm keeping the number of tracks reasonably low. Is it better to have each different segment on its own track, performing some sort of processing on the tracks as a group before mixing them down to, say, a music track and an ambiance track?
And, yes, I'm experimenting with all of this on my own as well. I've made a number of mundane recordings in different environments, trying to intuitively learn what levels work best where, and what I can expect to clip. I'm having a hard time getting a loud recording, though, and usually have to crank the volume to hear well but cut it so JAWS/VoiceOver isn't overwhelming. I'm not sure if this is something I need to address in the recording itself--not setting levels conservatively and risking the possibility of clipping--or if there's a certain series of processing steps I might apply and an order in which to perform them that might give a bit more volume without me worrying about whether or not I'm clipping. I'm getting closer, but I'm not quite there.
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