What is Audio Format Converter?

A free audio format converter that converts your files between MP3, WAV, OGG, FLAC, AAC, M4A, and Opus. Powered by ffmpeg.wasm, so your files stay completely private.

Seven output formats are available: MP3, AAC, OGG Vorbis, M4A, and Opus (all lossy, with selectable bitrates from 64 to 320 kbps), plus FLAC (lossless compressed) and WAV (uncompressed PCM 16-bit). MP3 and OGG can encode in VBR (variable bitrate) for better quality at the same average size. Quick presets — High-quality music, Podcast voice, and Ringtone — set the format, bitrate, sample rate, and channel mode in a single tap. A loudness normalization toggle re-levels each file to a platform target (Spotify and YouTube at -14 LUFS, Podcast at -16, Broadcast at -23), and an optional trim window keeps just the clip you want before converting. WebM files are also accepted as input, and you can queue several files at once for batch conversion. Sample rate (8 kHz to 48 kHz) and channel mode (mono or stereo) are exposed too, so you can target a specific delivery spec without leaving the page. Encoding runs in ffmpeg.wasm so source files never leave your device.

How to use

1. Upload one or more audio files (MP3, WAV, OGG, FLAC, AAC, M4A, or WebM). Drag-and-drop a whole folder of tracks or use Add more files to keep building the queue.
2. Pick the output format and, for a lossy codec, set the bitrate and choose CBR or VBR. Optionally set a sample rate, downmix to mono, normalize loudness to a platform target, or trim to a clip before encoding.
3. Click Convert. Each file is encoded in turn — download individual tracks from the list, or grab everything as a single ZIP once the queue is finished.

When to use

• Turning a studio FLAC master into a 320 kbps MP3 ready for Spotify or Apple Music upload.
• Replacing an M4A voice memo from an iPhone with a WAV that a Windows editor can open.
• Converting an OGG asset from a game engine into AAC for a YouTube background track.

Result

A musician receives a FLAC master from their studio but needs MP3 for streaming platforms. They upload the FLAC, select MP3 at 320kbps, and get a high-quality MP3 in seconds.

FAQ

Lossless to lossy versus lossy to lossless: what actually happens?

Going from FLAC to MP3 throws away frequencies the encoder considers inaudible (a real, permanent quality drop). Going from MP3 to FLAC packages the already-lossy audio into a lossless container, which keeps current quality but doesn't restore what was lost.

Can I change the sample rate or downmix to mono?

Yes — both controls live above the Convert button. Pick from 8, 22.05, 44.1, or 48 kHz, or leave Keep original to let FFmpeg copy the source rate. Channel mode offers Keep original, Stereo, or Mono; mono roughly halves the file size for speech recordings.

Why is the first conversion slower than later ones?

The ffmpeg.wasm engine is roughly 30 MB and only downloads on first use. After that it stays in browser memory for the session, so converting the second file feels almost instant compared to the initial load.

How big can the input file be?

Practically up to a few hundred megabytes on a desktop with plenty of RAM, but the limit is your device's memory. A full album lossless rip might struggle on a phone. If you hit a crash, split the file or use the desktop browser.

Does AAC inside an MP4 container get exported correctly?

Both work. Pick M4A and the output is wrapped in the MP4/iPod container with a .m4a extension — exactly what iTunes, the iPhone, and most podcast apps expect. Pick AAC and you get the same AAC-LC audio as a raw .aac stream instead. Either way the encoding is standard AAC-LC that VLC and modern Android play directly.

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