Convert AAC to MP3
Convert raw .aac recordings and stream captures into a 192 kbps MP3 that plays and seeks properly in almost any player.
Convert raw .aac recordings and stream captures into a 192 kbps MP3 that plays and seeks properly in almost any player.
A file ending in .aac is usually a bare AAC stream (often called ADTS) with no MP4 wrapper around it. That is different from M4A, which puts the same codec inside a container with an index and tags. Bare AAC files come from some voice recorder apps, saved internet radio streams, and tools that pull the audio track out of a video without re-encoding it.
Because there is no index, some players show the wrong length, can't seek properly, or refuse to open the file at all. An MP3 made here, encoded by LAME at 192 kbps constant bitrate, avoids those problems. Sample rates up to 48 kHz stay as they are, and mono or stereo is preserved; surround AAC is mixed down to stereo.
| AAC bitrate | Size per minute | Roughly fits in 50 MB | 192 kbps MP3 per minute |
|---|---|---|---|
| 64 kbps | 0.48 MB | about 1 hour 45 minutes | 1.44 MB |
| 96 kbps | 0.72 MB | about 70 minutes | 1.44 MB |
| 128 kbps | 0.96 MB | about 50 minutes | 1.44 MB |
| 192 kbps | 1.44 MB | about 35 minutes | 1.44 MB |
AAC is a lossy codec. To make an MP3, the audio is decoded and then compressed again with a different lossy codec, so there is a second round of loss. For a source at 128 kbps or higher, a 192 kbps MP3 normally sounds indistinguishable in everyday listening. What the conversion cannot do is make a poor recording sound better.
AAC is more efficient than MP3 at the same bitrate, and the output here is fixed at 192 kbps. So unless your AAC was already at 192 kbps or above, the MP3 will be bigger than the file you uploaded. A 64 kbps radio recording, for example, grows to about three times its size. That is the cost of compatibility, not a sign of improved quality.
Low-bitrate streams often use HE-AAC, which rebuilds high frequencies on playback from a small amount of side information. When decoded for conversion, that reconstructed treble becomes part of the MP3, so it should sound broadly similar to what you heard, including any of HE-AAC's typical artifacts.
.aac, destined for a device that doesn't recognize the extension.If the source showed an odd length, check the MP3's duration after converting. Since the audio is fully decoded and rewritten, the new file reflects the actual audio, but a stream that was cut off mid-recording may end abruptly.
Raw AAC streams carry little or no title and artist information, and whatever exists is not guaranteed to reach the MP3. Plan on naming and tagging the result yourself.
Files from iTunes or iPhones usually end in .m4a; for those, the M4A to MP3 page covers the details. DRM-protected purchases can't be converted in either case.
Uploaded AAC files are deleted the moment conversion succeeds, and finished MP3s are removed within about an hour. For editing-friendly output, try AAC to WAV or browse the AAC converter.
| AAC | MP3 | |
|---|---|---|
| Full name | Advanced Audio Coding (ADTS stream) | MPEG-1/2 Audio Layer III |
| Compression | Lossy | Lossy |
| Developed by | MPEG (Fraunhofer, Dolby, Sony, Nokia and others) | Fraunhofer IIS / MPEG |
| Introduced | 1997 | 1993 |
| File extension | .aac | .mp3 |
| Typically used for | Streaming, broadcast, audio from recorders and apps | Music libraries, podcasts, everyday listening |
| Plays on | iPhone, Android, modern browsers and most media players | Virtually every phone, computer, browser, car stereo and media player |
| Our output | AAC-LC, 192 kbps, raw ADTS stream | LAME MP3, 192 kbps CBR, stereo |
Both usually contain the same AAC codec. An .aac file is typically a raw ADTS stream with no container, while an .m4a file wraps the audio in an MP4 container with an index and tags. That missing index is why raw AAC files sometimes show the wrong duration or seek poorly in some players.
No. At the same bitrate, AAC is generally the more efficient codec. Converting AAC to MP3 is a lossy re-encode, so it adds a small amount of loss rather than improving anything. The reason to convert is compatibility with devices and software that handle MP3 but not AAC, not better sound.
The MP3 output is always 192 kbps, which is about 1.44 MB per minute. Many AAC recordings and radio streams use 64 to 128 kbps, so the MP3 ends up larger than the source. A 64 kbps recording roughly triples in size. The extra bytes come from the fixed bitrate, not from added detail.
The limit is 50 MB per file, not a time limit, so it depends on the bitrate. At 128 kbps that is about 50 minutes of audio, at 96 kbps about 70 minutes, and at 64 kbps well over an hour and a half. Longer recordings need to be split into parts before uploading.
No. There are no settings: every MP3 is LAME 192 kbps CBR, with the sample rate kept up to 48 kHz and mono or stereo preserved. Surround audio is mixed to stereo. Trimming, volume changes and tag editing are not available, so do those in an audio editor before or after converting.
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