Youtube hosts lots of music transcoded using faac and quality isn't great. As popular tracks are usually uploaded by multiple users, one might reduce encoding noise by averaging waveforms of duplicates (can be done either in client app or intermediate server). Noise std should decrease as 1/sqrt(N:). Do you guys think this would work?
Youtube hosts lots of music transcoded using faac and quality isn't great. As popular tracks are usually uploaded by multiple users, one might reduce encoding noise by averaging waveforms of duplicates (can be done either in client app or intermediate server). Noise std should decrease as 1/sqrt(N:). Do you guys think this would work?
If the sources are all identical rips from CD and the encoder/settings are all youtubes own, they might be all identical. In that case, averaging should give you nothing.
If different sources are used, and/or different encoders are used, you might have some improvement, but any phase-error would cause misalignement. In fact, there are many perception flaws that a lossy encoder might use to reduce the number of bits (without large audible errors) that might cause problems when averaging like you suggest.
-k
1/sqrt(N:) only applies to uncorrelated noise. I suspect that a lot of the noise added by lossy encoding is not entirely uncorrelated.
I hope this question is of a purely academic nature, because of course the right answer is if you want the music then buy your own copy.
I strongly suspect knutinh is right... Phase/timing differences in the different lossy encoders will probably cause weird
comb filtering (http://en.wikipedia.org/wiki/Comb_filter) effects. (This will
definitely happen if you mix two different
analog recordings of the same-exact performance.)
It's EASY to try!!! Just MIX the files and then normalize to bring-down volume. Do you have an audio editor? If not, Audacity is FREE!
Mixing is done by addition and volume adjustments are done by multiplication (up) or division (down), so this
is an averaging process.
Mixing two full-volume files will usually push the level above 0dB (the "digital maximum"). Most audio editors can handle levels above 0dB, but you should bring the levels down before saving to prevent clipping/distortion in the saved format. And of course, if you re-save in a lossy format, that's an additional lossy-compression step.
I hope this question is of a purely academic nature, because of course the right answer is if you want the music then buy your own copy.
Good point... If you find more than one copy of a recording, it's probably a copyrighted commercial release.