Chris, I'm curious. When you say that when you convert to wav, you already hear a noticeable loss... what are you comparing it to? Is the source also already on the computer in another format? In other words, how do you get your audio program into the computer initially? Microphone? Line-in to soundcard? Some other way? Or are you comparing your program on tape vs your program on the computer?
See the thing is this: wav is a lossless format. That means all of the original program material is there. Although there is a way to "compress" wav files, that generally is not the manner in which wav used, and the most common wav files are lossless and compression free. The problem with wav of course is file size. A 5 minute program could be 40mb. Flac is also a lossless format which is why it sound better than MP3 (and other lossy) formats. But unlike wav, it is compressed so you get the best of both worlds, lossless but small file sizes.
So the question is whether your initial quality loss is due to the manner in which the original program material is transfered to the computer or whether it is due to file conversion. Because you state that cutting the program to wav already degrades the program, I am leaning towards a economy hardware issue. Meaning cheap ADC (analog to digital converter). Remember, your audio program was originally analog and it had to be converted to digital in order for the computer to do anything with it. If the file the computer subsequently uses to "process" that information is already degraded, it only gets worse from there. No matter how much processing power your computer has, it is irrelevant if the digital information is already degraded during the conversion process.
Now, as far as chopping out.... one thing you need to understand is that once converted to digital from analog, ALL digital files are already cut. For example, if your audio was a perfect sine wave, it would be continuous wavy line that represents your audio signal. There would be no breaks in the line. However digital takes a snapshot view (sort of like drawing pictures in a booklet and flipping through the pages to create a moving image). So if this was audio, and it was being done the same way, the audio sample would be chopped up into evenly spaced "images". The coarser the pages or cuts, the more cut or loss in the music. The finer the pages or cuts, the smoother the music and less loss. I mentioned this in a previous post but in theory, if you chop it up fine enough, in theory the human ear isn't fast enough to hear the "lost" information. But I swear I can. In any event, if you look at any two adjacent digital snapshots, there is always space in between that is lost. Theoretically, the finer the cuts, the less space between the snapshots and the better the audio fidelity (less loss) but no matter how fine it is chopped up... a million times per second even, there will still always be information lost compared to analog since by definition, analog is continuous. That is the nature of digital, which is that the information is stored as snapshots in time.
So going back to the original topic. Is the problem with the transfer of audio from your original analog source to digital? Because once in the computer and rendered to a file, ANY file, it has already become digital.
and it just sounds flat, like someone stepped on it. It's very discouraging.
