MP3 to AIFF. Apple-native for Logic, GarageBand, Final Cut.

When you need AIFF instead of WAV

• Logic Pro — accepts both, prefers AIFF for in-project bouncing and Apple Loops. WAV works fine for drag-imports.
• GarageBand — same as Logic. AIFF is the native bounce format.
• Final Cut Pro — handles both, but FCP-generated audio assets are AIFF.
• Apple Loops authoring — Apple Loops Utility expects AIFF input for tagging metadata.
• Older Mac DAW projects — anything pre-Logic-9 era is more comfortable with AIFF.

What AIFF gives you that MP3 doesn't

• Lossless containment — no further generational loss if you edit and re-bounce.
• Sample-accurate editing — MP3 has frame boundaries that complicate exact cuts.
• Better DAW timeline behavior — uncompressed audio doesn't need decode-on-play.
• Compatibility with Apple's loop/region time-stretching engines.

What it doesn't give you: better audio quality than the source MP3. Lossy compression is one-way.

Why the file gets ten times bigger for no extra quality

This surprises people, so it is worth stating plainly. A three-minute MP3 at 192 kbps is about 4 MB; the same audio as 16-bit 44.1 kHz stereo AIFF is about 32 MB. Nothing has been added. AIFF simply stores every sample as a raw number instead of a compressed description, so the detail the MP3 encoder discarded stays discarded — it is now just written out at full size.

What you gain is a format that will not lose anything further. That is the entire reason to do this: you are about to edit, and you want the edits and the re-export to be lossless even though the source was not.

AIFF or WAV?

They are the same idea with different byte order and different metadata conventions — AIFF is big-endian and came from Apple, WAV is little-endian and came from Microsoft. Both store uncompressed PCM, both are lossless, and every DAW written in the last twenty years reads both.

Choose AIFF when the destination is specifically Apple: Logic Pro, Final Cut, Apple Loops Utility, or an older Mac project that expects it. Choose WAV for anything else, including Audacity, Reaper, Ableton, Premiere and every hardware sampler. If you have no particular reason, WAV travels further.

Frame boundaries, and the real reason to convert before editing

An MP3 is not a continuous stream of samples but a sequence of frames, each covering 1152 samples, and you cannot cut between them without either re-encoding or accepting a small imprecision. Encoders also add a short padding at the start, which is why an MP3 and its source rarely line up sample-for-sample.

Converting to AIFF resolves all of that into a plain array of samples, so an edit lands exactly where you put it and a loop is seamless. If you have ever lined up an MP3 loop in Logic and found it drifting slightly, that padding is the reason.

Bit depth and sample rate

The output is 16-bit at the source's own sample rate — no resampling, because resampling an already lossy file adds a second unnecessary transformation. 16-bit is the right choice here: the source is an MP3, its noise floor is far above the point where 24-bit would matter, and the extra depth would only make the file half again as large. If your project runs at 24-bit, your DAW will convert on import without complaint.