QUOTE
Imagine one algorithm throwing out 80+% of the music and than another taking that and throwing more away in it's own process.
It's not quite
that bad, because a lot of that 80% is "nothing useful".
Here are a couple of analogies that don't involve lossy compression... Let's say you have a 44.1kHz, 16bit file ripped from a CD. Let's say you resample it to 96kHz, 24-bit. That new file is about 3 times as big, and it contains 3 times as much data, but there's no more
useful audio data in the bigger file. Now, let's take that 96/24 file and downsample back to 44.1/16... We can throw-away 2/3rds of the data without loosing
any real audio information.
Or, say you have a 100Hz tone sampled at 44.1kHz. If you resample to 4kHz, you are throwing-away 90 percent of the data, but you have lost no audio information... You can still
perfectly reproduce the 100Hz tone!
Back to lossy... Both MP3 and AAC are trying to throw-away the least audible information. If you transcode (or encode a 2nd time), the encoder doesn't "look for" more audio to throw-away. There isn't as much real audio data in the file the 2nd time, and it doesn't
need to throw-out anything (assuming the same bitrate).
Of course MP3 and AAC are not identical (they throw-away different information) and the encoders are not perfect, so there will be
some additional audio data loss. But at a high-enough bitrate, the additional loss might not be noticeable.
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Never transcode lossy to lossy... Always have a copy of the lossless files around to experiment with lossy encoding.
That's an excellent rule-of-thumb, but sometimes we don't have an uncompressed original... You may have purchased an MP3 from Amazon, or an AAC from iTunes, or ripped an AC3 file from a DVD, etc.