These WMA seem awfully large for garden variety tracks. Are they WMA Lossless? Are these tracks abnormally lengthy?

Yes conversion rate usually depends on the hardware (aside from length of the audio), but also the efficiency of the converting tool. If it was using LAME, it should be real quick. In my opinion, your converter is far too slow for modern technology, thus is either: (a) working with copy protected WMAs, (b) working on a slow system, or © poorly written. For conversion, I use Foobar2000. Last I checked, it plays and converts (unprotected) WMA lossless.

Audacity can open more than just MP3. As far as I know, it has always worked with uncompressed WAV, and the latest works with FLAC [HA-Wiki] [Homepage].

As for the rest of your post... I can't help you. I haven't done much with sharpening and cleaning up audio files.

It absolutely make no sense to transcode to mp3 for editing in Audacity- just decode the original file to wav.

You are somewhat mistaken about Audacity opening mp3 files. WMA and mp3 are both lossy compression formats. Audacity opens only uncompressed files. It can recognize some forms of lossy compression and decode them to wav in order to open them, but it has not, and can not, actually open the lossy compression files.

By converting from WMA to mp3, you are first decoding to wav, then encoding to mp3. This second step somewhat reduces the audio quality (or rather more than somewhat, depending on parameters). It takes perhaps twice as long as simply decoding the WMA to wav (and it makes the audio worse). Then Audacity has to spend a significant amount of time decoding the mp3 back to wav before it can start loading the file into itself.

If you simply set things up to decode the WMA to wav, then open the wav in Audacity, you will get the maximum audio quality possible from the source WMA, and you will spend the least amount of time converting between formats.

Undesired noise and the target content are often within different frequency ranges in my experience - at least with hidden mic recordings I've made in the past - but this is not always the case.  I've usually been able to use the equalizer as described to "bring forward", so to speak, the sounds I want to hear.  The few times this didn't work for me, I had to take additional steps, either with other DSP effects at transcode time or with an audio editor after converting the file to WAV.

I just recommend the EQ/transcode approach as it's easy, and is usually a good first step to try.  Just make sure you DON'T delete the source file in the process; in fact, make a secure backup of it before doing anything.  (I'm sure you've already thought of this and done it, but it's worth mentioning.)

Also, the method you need to use will differ based on whether you just need to make the desired content reasonably audible (e.g., for manual transcribing or legal evidence) or need to produce something more refined (e.g., for publishing).

From your original post, it seemed to me like you just needed the target sounds to be more clearly audible, hence my recommendation to use dBpowerAMP to adjust frequencies, normalize the volume and get it into an MP3 format in (essentially) one easy step.  If you need to refine the resulting output more thoroughly, then of course you'll need to take further steps using a waveform editor such as the afforementioned Audacity.

NOTE:
Before embarking upon my approach, you can test it to determine whether it'll even be worth trying.  Play the original sample in an audio player with an equalizer, and adjust the EQ as I described earlier.  If that doesn't work, play around with the sliders any way you like until the sounds you want to hear are emphasized to your taste.  Then set the EQ in dBpowerAMP similarly when you use it to transcode to MP3.  However, if there's no way you can equalize the recording during playback to bring out the target content to your satisfaction, then my idea will not be the one to use.

Wow - that's a challenge.  As you say, the noise level is very high and the speaker's voice is relatively quiet.

I used the transcode/DSP method described above using dBpowerAMP Music Converter, and uploaded the encoded result here.  (Also note the setting details in that post, as well.)

The noise is still prevalent, but not as much so as in the original recording, and I was able to pull the voice above most of the noise (I believe) well enough to transcribe.  If most of the rest of your recording(s) are at an equal or higher recording quality than this one, then this approach may work well enough for you to be feasible.  (It's main advantage being essentially one step to improve the voice clarity and transcode to MP3, as these are your main goals.)

Note that (to keep the uploaded file size reasonable) I had to transcode from MP3 to MP3, and you or she (I assume) had previously transcoded it from WMA to MP3.  That's lossy-to-lossy-to-lossy, so the sound quality may suffer additionally just from that.  If you use this method starting with the original WMA file(s), then you may get a slightly better result than the one I uploaded.  (At least it won't be any worse.)

Let me know what you think.

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