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Supports: MP3
This is a one-way MP3 hub: upload an MP3 and convert it to WAV, M4A, OGG, FLAC, AAC, or Opus. The most common reason people land here is MP3 to WAV — they need an uncompressed file to drop into an audio editor, burn to an audio CD, or hand to a DAW that won't import compressed audio. MP3 is a lossy format, so the honest framing matters: converting MP3 to WAV or FLAC gives you a file that won't degrade further and plays in more software, but it does not add back detail the original MP3 encoder threw away. Every Conversion runs on our servers with no sign-up and no watermark.
MP3's patents fully expired in 2017 (the last relevant US patent lapsed in April 2017, after which Fraunhofer ended its licensing program), so it is now royalty-free and decodes on essentially anything made in the last two decades. That ubiquity is exactly why people still need to convert it — the file you have plays everywhere, but the workflow in front of you wants something else.
.m4r and trimmed under 40 seconds, is what iOS accepts as a custom ringtone.| Target | Type | Typical size, 4-min track | Plays natively on | Best for |
|---|---|---|---|---|
| WAV | Uncompressed PCM | ~40 MB (CD quality) | Everything | Editing, CD burning, DAW import |
| FLAC | Lossless | ~25-40 MB | VLC, modern Android, most editors | Tag-rich archival of the MP3 you have |
| M4A (AAC) | Lossy | ~4-7 MB | iPhone, Mac, broad elsewhere | Apple library, ringtones, streaming |
| AAC | Lossy | ~4-7 MB | Broad; broadcast and web | Efficient delivery, YouTube-style audio |
| OGG (Vorbis) | Lossy | ~4-7 MB | Chrome, Firefox, Android, VLC | Games, open-source projects, Spotify-style |
| Opus | Lossy | ~3-6 MB | Chrome, Firefox, Edge, modern Android, VLC | Smallest size at good quality, voice |
Sizes assume a 192-256 kbps MP3 source. Because the input is lossy, none of these outputs will sound better than the original MP3 — uncompressed and lossless targets only stop further loss, and lossy targets should be set at or above the source bitrate so you do not compound it.
When the output is a lossy format (M4A, AAC, OGG, Opus), the bitrate you pick sets the quality ceiling. Going above the MP3's original bitrate wastes space without adding fidelity, since the data is already gone; going well below it adds a second, audible round of loss. A safe rule: match the source bitrate or stay within one step of it.
| Setting | What it controls | Sensible choice |
|---|---|---|
| Audio Quality Preset | Encoder effort and target rate in one click | Highest for music; Low/Lowest for voice to save space |
| Constant Bitrate (CBR) | A fixed rate the player can rely on | 192-320 kbps for music; 96-128 kbps for spoken word |
| Variable Bitrate (VBR) | Spends more bits on complex passages | Best size-to-quality trade for music delivery |
| Audio Sample Rate | Samples per second | Leave at Original (MP3 tops out at 48 kHz); use 22-24 kHz for voice |
| Audio Channel | Mono vs Stereo | Mono halves size for a single speaker; keep Stereo for music |
The MP3 format itself caps sample rate at 48 kHz, so an MP3 source never carries the 96 or 192 kHz data that high-resolution files do. Converting to WAV or FLAC will produce a higher-bitrate file, but it is just a larger representation of the same 48 kHz-or-lower signal. As an illustrative example, a typical 4-minute MP3 at 256 kbps (~7.5 MB) expands to roughly 40 MB as 16-bit / 44.1 kHz stereo WAV — about a 5x increase in size for zero increase in audible quality.
No. MP3 is lossy — its psychoacoustic encoder permanently discards audio data during the original encode, and nothing downstream can reconstruct it. Converting to WAV or FLAC gives you an uncompressed or lossless file that is better for editing and won't degrade any further, but it sounds identical to the source MP3 while taking up several times the space. The only way to get true high quality is to start from a lossless master (FLAC, WAV, or ALAC), not from an MP3.
WAV. Editors like Audacity, Adobe Audition, Logic Pro, and Pro Tools decode WAV with no overhead and let you cut, mix, and process sample-accurately. If you keep working in MP3, every export re-encodes and stacks new compression artifacts on top of the old ones. Convert MP3 to WAV at the start, do all your edits, then export to your final delivery format (MP3, AAC, or Opus) exactly once.
iOS ringtones use the .m4r extension, which is simply AAC audio in an MP4 container — the same thing an M4A is. Convert your MP3 to M4A, trim it to 40 seconds or less (iOS hides longer files from the ringtone picker), then rename the .m4a to .m4r before adding it to your library through Finder or iTunes. Text-tone clips must be 30 seconds or shorter.
Opus. Standardized by the IETF as RFC 6716, Opus delivers better quality per kilobit than MP3, especially below 128 kbps, and is excellent for voice. The trade-off is compatibility: Opus plays in Chrome, Firefox, Edge, modern Android, and VLC, but not in older car stereos, iPods, or some hardware players. If you need both small size and universal playback, MP3 at 192 kbps stays the safe choice; if the target is a modern phone or browser, Opus wins.
Usually yes for the common fields. MP3 stores metadata as ID3v2 tags (including embedded cover art in the APIC frame). The converter maps Title, Artist, Album, Track number, Year, and Genre to the destination format's tag scheme — Vorbis Comments for FLAC and OGG, iTunes-style atoms for M4A. Less common fields (custom tags, ReplayGain, timed lyrics) may not transfer; if metadata is critical, verify in a tag editor like Mp3tag after converting.
Yes. Drop every MP3 into the file list, choose the output format and quality once, and click Convert — each file is encoded independently with the same settings, so loudness stays consistent and per-file tags survive. Download tracks one by one or as a single ZIP. For a long DJ mix or lecture saved as one MP3, use the Trim option to slice out sections, or the dedicated audio cutter for multiple splits.
Leave Audio Sample Rate on Original in most cases. An MP3 is already at or below 48 kHz, and upsampling it to a higher rate adds no information — it just enlarges the file. For music, 44.1 kHz is the universal target; for spoken-word content you can downsample to 22 or 24 kHz with Mono channel to shrink the output substantially without hurting intelligibility.
This hub changes the container and codec (MP3 to WAV, M4A, OGG, FLAC, AAC, or Opus). If you want a smaller MP3 that stays an MP3, use compress MP3 to lower its bitrate. If your audio is the soundtrack of a video file, use the dedicated extractors such as MP4 to MP3, which demux the audio stream and re-encode it to your chosen format.