MP3 to AAC Converter

MPEG-1 Audio Layer III (MP3) is a lossy audio format used for broad compatibility for music, voice and podcast downloads. Files saved as MP3 usually come from MPEG / Fraunhofer. Before converting, check that the source is the highest-quality copy you have, because every later step can only preserve what is already there.

The target is Advanced Audio Coding (AAC), a lossy audio format chosen for efficient audio in mp4. Both formats are lossy, therefore this is a second-generation encode: keep the bitrate high enough to limit audible artifacts. The output keeps the original dimensions or sample rate unless you change them, so this step changes the encoding rather than the content.

Raw AAC/ADTS keeps the efficient AAC codec but drops MP4 metadata and chapter support. It is convenient for streaming pipelines that expect elementary audio streams. Channel layout is preserved, so a stereo source stays stereo and a mono recording is not upmixed. The sample rate is preserved unless you change it; resampling would add another processing step and is best avoided. File size stays close to the source because both sides use comparable encoding efficiency.

This conversion runs locally in your browser tab through Canvas or FFmpeg.wasm. The file is read from your device, decoded, re-encoded and handed back as a download; it is not uploaded to a conversion server. That keeps private recordings, client material and unreleased artwork off shared infrastructure.

After the download, play or open the result before deleting the original. Check the first and last seconds, confirm that transparency, rotation metadata or sample rate survived, and compare file sizes. If something looks wrong, retry with a higher quality setting rather than converting the output again.