Can a compressed file be uncompressed and then edited without damage?
Would using the compressor effect on the edit be damaging?
Would exporting either as WAV or MP3 matter as well?
Is this the right section to post this?
Thanks.
Topic moved to “Audio Processing”.
“Compression” has two different meanings in relation to audio.
There is “data compression”. This is when a file is made smaller. It may be “lossless” (like ZIP or FLAC compression), or “lossy” (like WMA, MP3 or Ogg). “Lossy” compression discards some of the less important data, but in doing so there will be some degree of sound quality loss. “Lossless” compression keeps all of the data, but squashes it up together to make the file smaller. “Lossy” compression is able to compress files much smaller than “lossless” compression is able to do.
The other sort of compression is "dynamic range compression. This has nothing to do with file size. This is an “effect” that reduces the range between loud sounds and quiet sounds. It is often used to increase the overall loudness of the audio.
I think I knew dat./yeah, I knew that… my questions weren’t really about that as much as about the editing of a compressed ‘song’.
About uncompressing and editing compressed songs/or sounds in general. (I tried to search the topic but found no thing really-). Like to find info–on in’s and out’s of that… Will search more–thanks though. 
OK, so that article is about “data compression” (making the file size smaller).
Lossy formats such as MP3, AC3, WMA and so on, always reduce the sound quality a bit when the audio is encoded.
Audacity always works with “uncompressed” audio. That means, to edit an MP3 (or other compressed format) and end up with an MP3, the original MP3 is “decoded” to give uncompressed data, then on export it is re-encoded. That (re-)encoding will cause some sound quality to be lost - it’s similar to making a photocopy of a photocopy - do it enough times and you end up with a page of grey smudge.
There are some programs that can do basic editing (like trimming the length of a track) without decoding, so there is then no need to re-encode. Thus these programs are able to retain all of the quality of the original MP3. These programs can only perform “basic” editing. Advanced processing (as can be done in Audacity) is not possible without decoding the audio data. Examples of “lossless” MP3 editing programs are MP3Split and mp3DirectCut
By the way, I disagree with their conclusions and think their methodology is seriously flawed 
Thanks Steve. Still trying to grasp… it seems like the least damaging way to fully edit a compressed song (which would involve decompressing it) would be to just not export it as compressed when done, Is this correct? But just decompressing it in itself is damaging as well, correct? Anyway that’s my experience. I’ve concluded it’s not worth it, as Koz stated in another post, ‘to not edit/record compressed’ period. Like one of the big rules in music recording/editing.
My iPod recordings come out 64Kbps AAC as well (very low-fi/must be mono I assume[?]). The only reason I know (some AAC stats), is because I uploaded a program [FREE] by anysoft/anvsoft called Any Audio Converter so I could convert from AAC to MP3. Which begs another question, do you think that does damage as well? I can’t seem to notice, hard to tell though, (by ear anyway). It does seem to be more noticeable/less quality when I go from AAC to WAV for some reason.
It also seems like converting to 128Kbps 2 channels MP3 (from the AAC) seems to sound best[per conversions]/even better than 320Kbps for some reason. I assume if the AAC is 64Kbps, increasing the Kbps really is not doing anything, but since I made it 2 channels I assume 128Kbps is the best choice there. Also I think the sample[right word?] rate of the AAC is 48kHz (could that be at 64Kbps?), so I keep that when converting to MP3, so at least I don’t degrade there.
Converting to WAV format should be close to perfect.
Converting to MP3, Ogg, WMA (apart from WMA lossless), AC3 or any other “lossy” format will lose some sound quality, which may or may not be noticeable.
iPods that have internal hard drives may have trouble with WAV files because there is so much data. WAV is an “uncompressed” format, and for standard CD quality there are 1411 kb of data each second that need to be read from the hard drive. Compare that with a typical MP3 which is usually between 128 and 250 kb per second. The large amount of data puts a lot more demand on the hard drive which requires more power. Not only will that drain the batteries more quickly, but (conjecture) could account for reduced playback quality.
I just use the iPod to record. After that I use a regular PC computer for all else. Apple is so tight I wouldn’t want to deal with that anyway (and all I have Apple is the iPod). Anyway The iPod Voice Memo records AAC 44Khz mono 64Kbps (QuickTime’s ‘Movie Inspector’ shows fuller properties for AAC than Windows I found out).
“Converting to WAV format should be close to perfect.” Oh 
--you sure about that? I was thinking maybe when you convert from one compressed to another, there’s less decompressing/degrading… Now must rethink it all again!
The worse part about recording with the iPod (which actually produces the best quality recording/compared to USB and mono jack mics I’ve tried so far [cheapo’s albeit])…anyway, the worst part is trying not to clip. Which you would never know until you convert to wave and check in Audacity. There actually is a meter on the iPod but hard to monitor while recording. Fun hobby anyway…thanks Steve.
QUESTION: When recording mono (WAV) should the sample rate be set to half, say 22kHz, if you want the final export as a 44kHz stereo?
No, it must be set as 44100 Hz.
I just use the iPod to record.
How? I keep meaning to blast enough free time to try it with mine, but haven’t yet. Did you get one of those breakout adapters and use a “computer microphone?” A regular microphone with another adapter? Headset?
Radio Shack 3013?
http://www.radioshack.com/product/index.jsp?productId=2102927
It’s common to think the resulting sound files are always compressed, but several people think the compression is different. That’s a shame because it limits what you can do in post production.
Koz
I use the “Voice Memo” feature, nothing else. It delivers (by email only[Apple’s tight and rude, won’t even talk to poor Microsoft]) AAC(M4A[M4A to Microsoft I guess?]) 44kHz 64kBps mono --yeah buddy --‘Hi-Fi’. Then I convert it to wave, edit it with Audacity. Next experiment is a thrift store USB mic, $4. Which I found out is uni-directional (which I didn’t even know what that meant previously), barely picks up the guitar when singing/playing live, weird effect there. Eliminates the guitar being too loud though.
I have a similar question, but it may be that I need to start a new thread…
I finished part one of an episode of my podcast, recording via Audacity and using a Blue Snowball mic. (I am VERY new to recording and editing sound files, and am learning as I go, so technical terms will make my eyes glaze over!) After I finished adding in music, and had tweaked a phone message re-creation and put that in, I used “Normalize,” "Compressor,’ and “Mix and Render,” all on their default settings and in that order. I had saved a lossless copy, but forgot to name it as such as I usually do, and so I think I wrote over it when I did the finalizing. I did not listen to the final copy, as I had listened to it through right before the finalizing and figured it wouldn’t change. But the re-creation phone message changed enough that the vocals are too faint to understand, which I only found out after publishing the episode. So now I would like to fix that one piece, but I can’t see how to do that without messing the whole project up or just starting over. Is it possible to edit that one piece without making the rest of the project sound like crap? I had put it through filters, but made no notes as to what I did, and I feel really stupid for not saving a non-compressed copy of the whole thing!