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cdp788

macrumors newbie
Original poster
Apr 6, 2004
14
0
About half of my music library is encoded as AAC while the other half is MP3... fine for my iPod... but I'm thinking of buying a small flash player for when I go the gym. These players only play mp3s (or wma, but who wants that?)...

I have a very general knowledge of DSP and can't seem to understand why converting a music file from one format to another would result in a loss of quality. If it's a digital source, and the sampling frequencies are the same (44.1 kHz), shouldn't the data transfer smoothly? I understand the danger in convert from one bit-rate to another, but assuming I keep everything at 128 bps, how could I lose quality and where (amplitude, frequency response, other?).

Thanks.
 
cdp788 said:
About half of my music library is encoded as AAC while the other half is MP3... fine for my iPod... but I'm thinking of buying a small flash player for when I go the gym. These players only play mp3s (or wma, but who wants that?)...

I have a very general knowledge of DSP and can't seem to understand why converting a music file from one format to another would result in a loss of quality. If it's a digital source, and the sampling frequencies are the same (44.1 kHz), shouldn't the data transfer smoothly? I understand the danger in convert from one bit-rate to another, but assuming I keep everything at 128 bps, how could I lose quality and where (amplitude, frequency response, other?).

Thanks.

Transcoding from AAC to MP3 isn't a bad thing, but AAC is subjectivley a nicer sounding codec than MP3 at lower rates (to my ears 160 AAC sounds better than 320 MP3), so the real issue is the quality.

At 128 the artifacts produced by both codecs are noticeable to trained or acute ears, but not to general listeners, particularly on the limited "ear-bud" type headphones. iTunes should convert the tracks without pain, but you may not like the sound of the MP3 version so much.
 
That would be true if it weren't for the fact that mp3 and aac are compressed formats. If you transfer from one lossless format, like AIFF, to another with the same sampling and such, you lose no information. Those are too big for practical use on music players though. AACs are smaller, but you throw out some information to get it that small. No big deal, it tends to be insignificant so at higher bitrates you can't tell the difference. The trouble is, you lose information when you turn something to an mp3 as well. So you lose information and sound quality twice, which makes it more noticeable.

On the other hand, if you're just going to be using this in the gym on a cheap playher you might not notice or care if the quality is less than pristine.
 
basically, the AAC and MP3 encoders throw out different parts of the data when they compress a file. so you're losing data A when you make an AAC, and if you then turn that into an MP3 you also lose data B. if you go from AAC to MP3, you lose twice as much data as just compressing a file into one of the formats. that's what affects sound quality in a noticeable way.
 
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