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| quote: | Originally posted by Kev K
Incidentally, I could play a mix Live using purely vinyl and record it onto my puter using an mp3 encoder and sample it @ 320k 48000hz stereo. I would get a really good recording BUT, when I burned it back to cd for other people to listen to, it would only be 44.1khz 16bit which is a serious drop in sound quality and or file size (less file size means less quality which is something we learn early in college). |
Sorry, but I have to correct you on this!
First, the important thing with MP3 is the bitrate. 320k really stands for 320kbps (= kilobits per second)! So we have 320000 bits per second in that MP3 file (or 327680 bits to be exact, cause 1k = 1024 when measuring bits in a computer, but that's another thing ). A file with these specs and 1 minute in length would be 60(secs)*327680(bits/sec)=19660800(bits) -> 19660800(bits)/8 = 2457600(bytes). Note that this is regardless of all the other specs like sampling rate (48000 Hz), quantizer rate (16 bit), channels (mono/stereo) or whatever, these specs just say what is being reproduced when decoding this MP3 file.
MP3 compresses the recorded sound to fit more music into less storage space. But it uses some algorithms to do this in such a way that you hear as less difference to the uncompressed sound as possible. To explain it in a simple way, it "cuts off" the parts of the sound which we humans can't even hear (certain frequencies).
With a higher bitrate, there is (obviously) more space for the compressed sound to be stored, so regarding ONLY MP3 files your statement "less file size = less quality" is correct, cause with MP3 compression, using the same algorithms:
less bitrate = less file size = more compression necessary = less sound quality.
But you can't compare this to WAV files, cause there is no compression on those! The important specs on WAV are sampling rate, quantizing rate and number of channels. Let's say we have a WAV with 44.1kHz sampling rate and 16 bit quantizer rate in stereo (your example). Then we have 44100 sound samples per second (=Hz), and for each sample there's a 16 bit storage space reserved to store the value of this exact sample. That means, if we use the sound file from above (1 minute length), we would have 60(secs)*44100(samples)*16(bits)*2(channels, stereo!)=84672000(bits) -> 84672000(bits)/8 = 10584000(bytes).
You see that the WAV file is over 4 times the size of the MP3 file, even despite the fact that we have "only" 44.1 kHz sampling rate in the WAV file compared to the 48 kHz in the MP3 file!
So if you burn that 320kbps mix of yours down onto a CD, the quality doesn't drop at all! However, of course, the quality won't improve either, since obviously the copy can only be as good as it's master recording. 
It's in fact the other way around:
If you have a CD, which was recorded directly from the source (say, a classical concert for instance), it's quality would degrade if you convert it to MP3 files on your computer! But with a high enough bitrate (common thought is that this is starting with 192kbps), you won't hear that lesser quality because of these compression algorithms that I explained earlier.
*phew*
Sorry for this long speech, but I've studied all those things and couldn't help myself... 
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The world we knew is gone.
The conditions which framed and taught us our standards have gone with it.
Our needs are now different, and our aims must be different.
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