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Is it possible to improve the qualities of an mp3 so that it is more close to WAV or Aiff file audio?

Also can you remove the aliasing?

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Are you trying to make an mp3 file sound better when converted to WAV/AIFF?

If so, there is nothing much you can do. I usually use iZotope Ozone's harmonic exciter to 'fill in' some of the gaps. It works ok, but obviously depends on the quality of the mp3.

If you are trying to convert a wave to mp3 and achieve better quality:

Higher the bit rate, bigger the file size. Most encoders/decoders can go to a maximum of 320kbps - which sounds pretty good to my ears. Although, encoding mp3s with a variable bit rate (VBR) can help achieve better quality with smaller file sizes.

Also some encoders (like the one in iTunes) offer other options (like filtering low frequency content, stereo mode, etc). A combination of these different settings could help achieve smaller file sizes at a better quality - depending on the source of course.

Most encoders also need some amount of headroom, even limiting to -0.5 dBFS can help avoid distortion.

Some good info here:

http://www.gearslutz.com/board/mastering-forum/349716-wav-mp3-conversion.html

For the myths on join stereo and some info on mp3 encoding:

http://harmsy.freeuk.com/mostync/

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  • Im trying to make an mp3 sound better..
    – Chris
    Aug 29, 2010 at 18:29
  • @Chris That's like saying you're trying to make a dollar out of 14 cents - can't be done.
    – Utopia
    Aug 29, 2010 at 19:48
  • Yeah... with mp3 being lossy, you're outta luck. Once something's missing, it can't be brought back (same reason you can't make a telephone recording sound like anything but a telephone recording).
    – Dave Matney
    Aug 30, 2010 at 1:27
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MP3 is lossy compression, so once the source material is compressed you're pretty much stuck with what's left. If you hear aliasing there isn't any getting away from it in my experience.

With that said, there are certainly ways to make very good mp3 encodings. The MP3 export from protools is one of the best sounding i've heard. I did a blind A/B of a bunch of MP3 codecs a while back, and the protools one was the winner by a mile - it beat out itunes, barba batch, and a couple of others.

Remember that things like content and bitrate are the keys in mp3 encoding. The higher the bitrate and the simpler the source material the better.

you'll hear artifacting more quickly on the reverb tails of a string quartet than you will on a close miked dry bulb horn for example. The human voice actually encodes very well.

As technology marches on the need for lossy encoding will continue to diminish. Its already been dramatically reduced in video games as compared to a few years ago. The same will happen with other broadcast and distribution methods as storage gets cheaper and pipes get wider.

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I agree with most that has been said so far.

I'd like to add that the reason you get aliasing can be explained by the Nyquist-Shannon Sampling Theorem, which addresses sample rate requirements for certain frequency bandwidths. Basically, you need twice the sample rate of a specific frequency to be able to capture it properly. So, if you want to grab 20khz, you'll need at least a 40khz sample rate. This isn't a problem with wav and aif files, only with compressed files. As the sample rate gets lower, you start ending up with harmonics from the frequencies that are getting cut off. These are usually referred to as "Foldover" frequencies. For this reason, you need an anti-aliasing filter.

Anyways, I won't write too much about it here. If you'd like more info about it, check out this link:

http://en.wikipedia.org/wiki/Nyquist%E2%80%93Shannon_sampling_theorem

Contains mathematical proofs and theory behind what's happening.

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  • @Colin, this part of the article sounds a bit sarcastic and very funny to me: "The theorem is commonly called the Nyquist sampling theorem, and is also known as Nyquist–Shannon–Kotelnikov, Whittaker–Shannon–Kotelnikov, Whittaker–Nyquist–Kotelnikov–Shannon, WKS, etc., sampling theorem, as well as the Cardinal Theorem of Interpolation Theory. It is often referred to as simply the sampling theorem."
    – Justin Huss
    Aug 30, 2010 at 0:08
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Hey Chris... by no means a solution to an ugly sounding .mp3, but in desperate situations I've used a small amount of reverb with mostly early reflections to mask some of the ugliness and re-introduce some width. Try some tight-sounding rooms, perhaps impulse responses from a studio's Iso booth or the like. J

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Izotope RX 8 has spectral recovery tools for restore low quality mp3 to wav Demo

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