Free online MP2 to AAC converter

A phone, editor, media library, or delivery system may accept AAC while rejecting MPEG Audio Layer II. Converting can also reduce file size at similar perceived quality, but it cannot recover detail removed by the original MP2 encoding.

What MP2 is

MP2 is MPEG Audio Layer II, a lossy audio codec usually stored as a .mp2 file. It is not MP3: MP2 is Layer II, while MP3 is MPEG Audio Layer III.

MP2 is still used in broadcast and legacy media workflows, particularly MPEG program and transport streams, DVB services, radio distribution, broadcast automation, and older encoders. It is widely supported by desktop players, but support in phones, consumer devices, and current editing software is less consistent.

What AAC is

AAC (Advanced Audio Coding) is a lossy codec used in mobile devices, streaming, video production, and modern media software. AAC-LC is the normal profile for music and general-purpose audio; it is generally more bitrate-efficient than MP2, especially at low and medium bitrates.

AAC can be delivered as an ADTS-framed .aac stream or inside a container such as .m4a, .mp4, or .mov. An .m4a file is usually preferable for music libraries because it can carry standard metadata, artwork, and reliable duration information; use raw .aac only when the receiving system explicitly requires it.

Convert the file

Upload the .mp2 file to 101convert.com, select AAC as the output format, start the conversion, and download the converted file. The service can be used free in a browser without installing software. Check the destination specification first: some systems require a raw .aac file, while others require AAC audio in an .m4a or video container.

For repeatable local conversion or exact control of output settings, use FFmpeg. This command creates a raw AAC stream:

ffmpeg -i input.mp2 -c:a aac -b:a 192k output.aac

This command places AAC in an M4A container:

ffmpeg -i input.mp2 -c:a aac -b:a 192k output.m4a

For a target that specifically requires 48 kHz stereo, set the sample rate and channel count explicitly:

ffmpeg -i input.mp2 -c:a aac -b:a 192k -ar 48000 -ac 2 output.m4a

Preserve the source sample rate and channel layout unless a device, broadcaster, or video project specifies different values. For stereo speech, AAC-LC at 96k to 128k is often sufficient; for stereo music, 160k to 256k is a typical range, with 192k as a practical starting point.

Quality and compatibility limits

This is a lossy-to-lossy transcode. MP2 is decoded and encoded again as AAC, so MP2 artifacts remain and AAC encoding can add new ones. Keep the original MP2 file, or preferably the uncompressed source, for archiving or future editing.

A higher AAC bitrate can reduce additional encoding loss, but it cannot restore frequencies or detail discarded by MP2. Test a short representative section when converting low-bitrate MP2, music with sharp transients, or material intended for further editing.

Raw .aac files commonly use ADTS framing and do not provide the metadata support expected by many music applications. Browser and device compatibility also depends on the container: a player that supports AAC in .m4a or .mp4 may not accept a standalone .aac file. Verify both the required codec and container rather than relying on the filename extension.