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Supports: AAC
AAC (Advanced Audio Coding) is the format that replaced MP3 as the default for modern audio. Standardized by ISO/IEC in 1997 as MPEG-2 Part 7 and folded into MPEG-4 in 1999, it squeezes better sound out of the same number of bits — a 96 kbps AAC stream is roughly as good as a 128 kbps MP3. That efficiency is why YouTube, Apple Music, Spotify, and digital broadcast all lean on AAC. The catch is reach: older car stereos, basic MP3 players, and a lot of legacy software still expect a plain .mp3. This converter takes an AAC file and re-targets it to whatever your destination actually accepts — MP3 for compatibility, WAV for editing, M4A to keep it in the Apple container, FLAC for a lossless archive, or OGG/Opus for open-format and low-bitrate use.
.aac from any source — phone recordings, ripped tracks, or audio demuxed from video.| Convert AAC to | Result type | Pick it when |
|---|---|---|
| MP3 | Lossy transcode | You need a file that plays on virtually any device or app made in the last 20 years |
| M4A | Lossless re-wrap* | You want to stay in the Apple/iTunes ecosystem and keep the original quality |
| WAV | Uncompressed | You are importing into Audacity, Audition, Logic, or Pro Tools for editing |
| FLAC | Lossless container | You want a tagged archive copy that is smaller than WAV |
| OGG (Vorbis) | Lossy transcode | You need a royalty-free format for games, Spotify-style apps, or open-source projects |
| Opus | Lossy transcode | You want the smallest possible file that still sounds good (voice notes, streaming) |
*AAC to M4A is lossless only when the audio is copied without re-encoding (the codec stays AAC). If you change the bitrate or re-encode, it becomes a lossy transcode like any other.
AAC and MP3 are both lossy: each one permanently discards audio data to shrink the file, so converting between them is a transcode, not a lossless copy. The difference is efficiency. AAC uses a more modern encoder — a pure MDCT filter bank, temporal noise shaping, and smarter stereo coding — so at the same bitrate it preserves more detail. Below 128 kbps the gap is clearly audible; AAC at 96 kbps holds up where MP3 at 96 kbps starts to smear. At 256 kbps and above, most listeners cannot tell the two apart in a blind test.
| AAC | MP3 | |
|---|---|---|
| Standard | ISO/IEC 13818-7 (1997), 14496-3 | ISO/IEC 11172-3 / 13818-3 |
| Compression | Lossy | Lossy |
| Typical bitrate | 128–256 kbps | 128–320 kbps |
| Sample rates | 8 kHz – 96 kHz | 8 kHz – 48 kHz |
| Quality at 96 kbps | Good | Noticeably weaker |
| Quality at 256 kbps | Transparent for most | Transparent for most |
| Container | M4A / ADTS | MP3 |
| Patent status | Licensed (Via LA pool) | Patents expired (2017) |
| Best for | iPhone/Mac, YouTube, streaming, broadcast | Universal playback, car stereos, podcasts |
The practical takeaway: convert AAC to MP3 when the destination matters more than squeezing out the last few percent of quality — an old head unit, a fitness watch that only reads MP3, a website upload that demands it. If the destination already supports AAC, converting to MP3 only trades quality away for nothing.
When you transcode AAC to another lossy format, the output bitrate sets the ceiling on quality. Because your source is already lossy, going higher than the source bitrate cannot add detail back — it just makes a bigger file. Use this as a target reference.
| Bitrate | What it is good for |
|---|---|
| 64–96 kbps | Speech, podcasts, voice memos — small files where music fidelity is not the point |
| 128 kbps | Baseline music quality; fine for casual listening and most YouTube audio |
| 192 kbps | Very good music; a safe default when converting AAC to MP3 for general use |
| 256 kbps | Near-transparent; matches what Apple Music streams in AAC |
| 320 kbps | The MP3 ceiling and a transparent target for most listeners |
For lossless targets (WAV, FLAC, ALAC in M4A) bitrate is not a setting — they store the decoded audio in full. Just remember that decoding a lossy AAC file into a lossless container does not recover the data AAC already threw away; it only stops further loss from that point forward.
Yes, slightly, because both formats are lossy and the audio gets re-encoded. AAC has already discarded some data, and MP3 discards more on top of that, so a small amount of generation loss is unavoidable. To keep it minimal, set the MP3 output at or above the source bitrate — 192 or 256 kbps is a good safe target. Going higher than the source does not improve anything; the result can only ever be as good as the original AAC, never better.
It can be. M4A is a container (MPEG-4 Part 14), and AAC is the codec that most commonly lives inside it. If the conversion simply re-wraps the existing AAC stream into the M4A container without re-encoding, no audio data is lost — it is the same audio in a different wrapper, which is why this conversion is fast. If you change the bitrate, sample rate, or codec during the conversion, it stops being a pure re-wrap and becomes a lossy transcode like any other.
To stop further quality loss, not to recover lost quality. A WAV or FLAC made from an AAC file is bit-perfect from that point on — useful when you need to edit in a DAW (which prefers uncompressed WAV) or archive a copy that won't degrade if you process it again. What it cannot do is rebuild the detail AAC already removed during its original lossy encode. If a true lossless master exists, always convert from that instead of from the AAC.
Basic metadata — title, artist, album, track number — transfers in most conversions where the target format supports tagging, including MP3 (ID3), M4A, FLAC, and OGG. WAV is the exception: its RIFF container has no standard tagging scheme, so a WAV export typically loses embedded tags and cover art. If keeping a tidy, taggable library matters, FLAC or M4A is a better lossless target than WAV.
For music, 44.1 kHz (CD standard) is the universal choice and matches what most AAC files already use. Apple and broadcast workflows often use 48 kHz because it lines up with video. The safe move is to leave Audio Sample Rate on "Original" so the converter keeps the source rate — upsampling a 44.1 kHz file to 96 kHz adds no information and just inflates the file size.
Opus. Standardized by the IETF (RFC 6716), Opus outperforms MP3 at low bitrates and holds its own against AAC up to around 96 kbps, so it gives the smallest file for a given quality level. The trade-off is reach: Opus plays in Chrome, Firefox, Edge, modern Android, and VLC, but not in many older car stereos, basic MP3 players, or some standalone hardware. If you need both small and universally playable, MP3 at around 128 kbps is the safer pick.
Yes. Use the Trim control in Advanced Options to keep only a start-to-end window of each file, and use "Specific file size" to make the output land near a byte target — handy for fitting an attachment limit. For size-driven batch jobs or repeated trims, the dedicated Audio Compressor and Audio Cutter give you more control over the output.
As an illustrative example, a 3-minute AAC track at 256 kbps converted to MP3 at 192 kbps comes out around 4.3 MB, versus roughly 30 MB if you decode the same track to uncompressed WAV. FLAC of that track lands near 18–22 MB depending on the music — lossless but far smaller than WAV. The pattern holds generally: lossy targets (MP3, OGG, Opus) produce the smallest files, lossless containers (FLAC, ALAC) sit in the middle, and uncompressed WAV/AIFF is always the largest.