Convert WAV to MP3
Convert uncompressed WAV audio to MP3. Reduce file size up to 90% with no perceptible quality loss. Free, in your browser.
.wav · up to 100 MB
Why convert WAV to MP3
Uncompressed audio, compact file size
90% size reduction
From 50 MB WAV to 5 MB MP3: libmp3lame at 192 kbps with quality virtually indistinguishable from the original.
Total privacy
Your audio never leaves your device. FFmpeg.wasm processes everything locally, no uploads.
Compatible everywhere
MP3 works in any player, platform, streaming service, or device in the world.
Instant after first load
The FFmpeg engine downloads once and is cached. All subsequent conversions start immediately.
How it works
Three steps, no hassle
Select your WAV file
Drag or select your .wav file — a recording, music track, sound effect, or production audio. No registration required.
MP3 compression with libmp3lame
FFmpeg.wasm uses libmp3lame to encode the audio to MP3 at 192 kbps. Everything happens on your device, no uploads.
Download your MP3
The resulting MP3 is up to 90% smaller than the original WAV with virtually indistinguishable quality. Ready to share or play.
FAQ
Got questions?
WAV is uncompressed audio (PCM), meaning it contains all original audio data. MP3 is a lossy format that discards frequencies the human ear doesn't easily perceive. At 192 kbps or higher, the difference is practically inaudible to most listeners under normal conditions. Only in high-fidelity listening with high-end headphones and in direct comparison might any difference be noticed.
The reduction is dramatic: a 50 MB WAV file becomes approximately a 5 MB MP3 at 192 kbps — a 90% reduction. This difference exists because WAV stores each audio sample uncompressed (typically 16-bit at 44,100 Hz), while MP3 applies the psychoacoustic model to discard auditorily irrelevant data.
Keep the original WAV for music production (mixing, mastering), long-term archival recordings, and any process where you'll edit or re-export the audio. Each time you re-export an MP3, quality degrades slightly by applying lossy compression on top of lossy compression. For final use (listening, distribution, podcasts), MP3 at 192 kbps is completely adequate.
Bitrate determines how much data per second is used to represent the audio. 128 kbps is acceptable for voice and podcasts; 192 kbps is the standard for quality music (practically indistinguishable from the original for most listeners); 256 kbps is for audiophiles or professional music distribution; 320 kbps is the MP3 maximum, equivalent to CD quality for most ears. This tool uses 192 kbps as the default.
MP3 supports ID3 tags for storing metadata such as artist, title, album, year, and cover art. If your WAV file contains metadata (some DAWs write them in the header), FFmpeg will attempt to transfer them to the MP3. However, metadata in WAV is not as standardized as in MP3, so there may be variations.
Yes. The tool preserves the original sample rate from the WAV file. If your WAV is 44,100 Hz (CD standard) or 48,000 Hz (video/broadcast standard), the MP3 will maintain that same rate. This is important for maintaining compatibility with video projects or productions with audio synchronization.
WAV to MP3: WAV format history (Microsoft/IBM 1991), PCM uncompressed audio, MP3 compression algorithm (psychoacoustic model), and music production workflows
The WAV (Waveform Audio File Format) format was jointly developed by Microsoft and IBM in 1991 as part of the RIFF (Resource Interchange File Format) specification for Windows 3.1. WAV stores audio in PCM (Pulse-Code Modulation) format, the most direct digital representation of analog sound: amplitude samples taken at regular intervals (typically 44,100 times per second for CD quality) encoded as numeric values. This absence of compression guarantees complete fidelity to the original audio, but at the cost of large files: one minute of stereo audio at 44,100 Hz and 16-bit depth occupies approximately 10 MB.
MP3 (MPEG-1 Audio Layer III) was developed at the Fraunhofer Institute for Integrated Circuits (IIS) in Germany during the 1980s, with fundamental contributions from Karlheinz Brandenburg. The algorithm was patented in 1989 and the MPEG-1 standard published in 1993. The technical key of MP3 is the psychoacoustic model: a mathematical system that models the limitations of human hearing to identify which audio information can be discarded without being perceptible. Auditory masking (a loud sound hides nearby weaker sounds in frequency) and temporal masking (a loud sound masks sounds just before and after it) allow 80-90% of audio data to be eliminated with minimal perceptual loss. The last MP3 patents expired in 2017, making the format completely free.
WAV-to-MP3 conversion is fundamental in music production and audio distribution workflows. DAWs (Digital Audio Workstations) like Ableton Live, Logic Pro, and Pro Tools work internally with WAV or AIFF for maximum quality during production. The WAV file is the exchange format between studio and master. But for distribution — streaming platforms (Spotify accepts WAV/FLAC but serves MP3/AAC), podcasts, social media, or simply sharing music — MP3 is the universal standard. The libmp3lame library, used by this tool, is the same one used by FFmpeg, YouTube, and virtually all open-source MP3 encoders in the world, guaranteeing the highest quality compression available.