When you import a lossy audio file into editing software, that file is decoded into an audio waveform. This is a "lossless" waveform in that it's not stored in a compressed format, but it's not the same as the original waveform that your lossy file was approximating.
So there are already three concepts involved:
- The original waveform, which is to say, the "full-quality," unmodified waveform
- The lossy encoding of the original waveform (not a waveform itself, but it decodes into one that's almost the same as the original)
- The waveform that gets decoded from (2)
When we talk about "quality" here, we mean "how close we are to the original waveform." So waveform (1) is the highest-possible quality, and everything that approximates that involves some loss of quality.
When you have your decoded waveform in your software, and you re-encode it to another lossy format, you incur a quality loss. Your new encoded file is an approximation of (3), no matter what quality you choose. But (3) was an approximation of (1). So no matter what you do, you're incurring a quality loss every time you re-encode with a lossy format, even if the amount of loss is marginally smaller (in other words, even if you choose a 'higher quality' lossy encoding than was chosen for the first encoding).
You absolutely should be worried about this if you're going to be editing and re-editing your files. If you want to preserve quality between editings you should be using a lossless format like wav, aiff, or FLAC.
You can test this by opening a lossy file and re-encoding it with something lossy, and then re-opening and re-encoding many times. It won't be long before the errors introduced are pretty obvious (though to be fair I've never run this test)