Audio Converter & Compressor
Convert and compress mp3, aac, ogg, opus, wav, and flac audio in your browser via ffmpeg.wasm. Pick a target format, bitrate, sample-rate, and channel mode, and see before / after waveforms side by side.
Interactive tool
ffmpeg.wasm engine loads on first run (~32 MB)
The engine is cached for the rest of your session. Every transcode runs entirely in your browser: files never leave your device.
Threading: detecting…
Drop an audio file here
Up to 250 MB / 5 minutes. The file never leaves your browser.
What is the Audio Converter & Compressor?
A free, in-browser audio converter and bitrate compressor. Convert between mp3, aac, ogg, opus, wav, and flac, drop the bitrate to save bandwidth, downmix to mono, or resample: all through ffmpeg.wasm. No upload, no signup.
How to Use the Tool
- 1Drop an audio file (mp3, aac, ogg, opus, wav, flac)
- 2Pick Convert (switch format) or Compress (lower bitrate, keep format)
- 3Tune bitrate, sample-rate, and channel mode
- 4Click Process: ffmpeg.wasm loads on first run, then is cached
- 5Compare the before / after waveforms and download
Key features
Six target formats
mp3, aac, ogg (Vorbis), opus, wav, flac: most-compatible codec per container
Convert and compress modes
Switch format outright or shrink in place by lowering bitrate / channels
Bitrate slider (24-320 kbps)
For lossy codecs: instantly preview the size hit and savings
Sample-rate dropdown
Resample to 8 / 16 / 22.05 / 32 / 44.1 / 48 kHz for voice / podcast / archival
Mono / stereo / preserve
Downmix to mono for podcasts and voice memos to save half the bytes
Before / after waveform thumbs
See the dynamic range of the source vs the encoded output side by side
Multi-threaded ffmpeg.wasm
Uses SharedArrayBuffer when available for faster encoding
100% private
No upload, no logs: the audio stays in your browser
Why Use This Over an Online Converter?
Online audio converters upload your file, often watermark or rate-limit the output, and don’t let you tune sample-rate or downmix. Doing the encoding locally is private, free of watermarks, and gives you the same level of control as a desktop ffmpeg install.
Common use cases
- Shrink a 50 MB lossless wav podcast cut down to ~5 MB mp3 for distribution
- Convert a flac archive to opus for low-bandwidth syncing to a phone
- Downmix a stereo voice memo to mono and halve the file size
- Re-encode an mp3 at 64 kbps mono for transcript-quality voice notes
- Convert an old aac file to mp3 to work in legacy editors
Codec cheatsheet
- mp3: universal compatibility, dated psychoacoustic model. Good baseline.
- aac, better quality per kbps than mp3, plays everywhere modern.
- opus: best-in-class quality at 32-96 kbps, perfect for podcasts and VoIP.
- ogg (Vorbis): open-source mp3 alternative, great for games / web.
- wav: uncompressed PCM, big but lossless, ideal for editing.
- flac: lossless compression, ~50% smaller than wav, perfect for archival.
Picking a bitrate
- 32-64 kbps: voice only (mono), opus only. Save the bytes.
- 96 kbps: typical podcast, surprisingly good with opus.
- 128 kbps: historical mp3 default, transparent on most consumer ears.
- 192 kbps: quality reference for music streaming.
- 256 kbps: Apple Music quality, near-transparent.
- 320 kbps: mp3 ceiling, archival-grade lossy.
When to compress vs convert
Compress mode keeps the source format and just shrinks the bitrate / channels, perfect for "I have a 9 MB mp3 that I want under 5 MB". Convert mode is for switching format outright (e.g. flac → mp3).
Tips & best practices
Use opus for podcasts at 64 kbps
opus at 64 kbps mono is roughly indistinguishable from a 192 kbps mp3 for spoken word. Massive bandwidth savings.
Resample to 22.05 kHz for voice
Human voice rarely needs more than 22.05 kHz sample-rate. Halving from 44.1 saves another 30-40% without audible loss.
flac for archival, opus for sharing
Keep the master in flac, ship at opus 96 kbps. You retain a lossless source forever.
Privacy & security
Encoding runs entirely in ffmpeg.wasm inside your browser. The source audio is never uploaded, logged, or stored. Engine artefacts are served from our own origin (not a third-party CDN).
Related tools
- Noise Generator
Generate white, pink, brown, blue, or violet noise in your browser: up to 30 minutes. Pick a duration, sample rate, channels, and loudness; export as wav, mp3, aac, ogg, or opus. Perfect for sleep, focus, ASMR, UGC voice-over beds, and tinnitus masking. No upload, no signup.
Audio Tools
- Audio Trimmer
Trim mp3, aac, ogg, opus, wav, and flac audio in your browser with ffmpeg.wasm. Drag the waveform thumbs to set start and end, add optional fade-in / fade-out, and keep the source format. No upload, no signup.
Audio Tools
- Audio Looper
Loop a short audio clip into a long one (up to 30 minutes) in your browser. Pick a target duration, get a seamless crossfade by default, or use byte-perfect Fast mode for pure noise sources. Powered by ffmpeg.wasm. No upload, no signup.
Audio Tools
- Video Compressor
Shrink mp4, webm, and mov videos right in your browser with ffmpeg.wasm. Choose a target preset (web 720p / 1080p, Discord 25 MB, Twitter 512 MB) or set a custom H.264 CRF: your files never leave the device.
Video Tools
- Background Remover
Remove solid or green-screen backgrounds with chroma keying. tolerance and feather in your browser. Sample key from a pixel or choose a color.
Image Tools
- JSON Formatter & Validator
Format, validate, and beautify JSON data instantly in your browser
Developer Tools
Frequently Asked Questions
Which formats are supported?
Input: mp3, aac, m4a, ogg, opus, wav, flac (plus anything ffmpeg.wasm can decode). Output: mp3, aac, ogg (Vorbis), opus, wav (PCM 16-bit), flac. The tool auto-pairs the right codec for each target.
Convert vs Compress: what is the difference?
Convert mode switches the format outright (e.g. flac → mp3). Compress mode keeps the source format and just lowers the bitrate / channels / sample-rate. Compress is best when you want a smaller file in the same format.
What bitrate should I use?
64 kbps is plenty for voice in opus. 96-128 kbps is great for podcasts. 192 kbps is the music sweet spot for mp3 / aac. 256-320 kbps is near-transparent. wav and flac ignore the bitrate slider. They are lossless.
Does my file get uploaded?
No. Encoding runs locally via ffmpeg.wasm. The engine artefacts are served from the same origin under /ffmpeg/<version>/.
How big a file can I process?
The soft cap is 250 MB / 5 minutes to protect mobile browsers from out-of-memory crashes. Larger files generally work on desktop.