How to convert wav to mp3 — free, browser-based, no upload
- Step 1Open the WAV to MP3 tool — Load the wav-to-mp3 tool. FFmpeg's WebAssembly engine fetches on first use (a few MB, cached afterwards) — everything after that runs offline in the tab.
- Step 2Drop your WAV file — Drag a single
.wavonto the dropzone or click to browse. The tool reads it into browser memory; nothing is sent to a server. It accepts standard PCM WAV at any sample rate and bit depth (16, 24, or 32-bit float), mono or stereo. - Step 3Choose a target bitrate — Pick from
320 kbps · best,256 kbps,192 kbps · default,128 kbps · podcast, or64 kbps · voice. This is the only option for this tool — it sets FFmpeg's-b:aconstant bitrate. Higher = better fidelity and larger file. - Step 4Encode — Click to run. FFmpeg transcodes WAV PCM to MP3 with
libmp3lameentirely in your browser. A short recording finishes in under a second; a long studio WAV scales with duration since WebAssembly runs on the CPU. - Step 5Download the MP3 — The encoded MP3 is offered as a download with the same base filename and a
.mp3extension. Tags and cover art from the source WAV are written into ID3v2.3. - Step 6Verify it plays and is tagged — Open the MP3 in any player to confirm length and audio. If you need to add or fix title/artist/album, follow up with the id3-editor tool; if you need a smaller file for a specific platform, see the audio-compressor tool.
Bitrate options and what they're for
The five bitrates exposed in the tool, mapped to FFmpeg's -b:a value and a typical use. All are constant bitrate (CBR) MP3 via libmp3lame. File-size figures are approximate per minute of stereo audio.
| Tool option | FFmpeg -b:a | ~Size / minute (stereo) | Best for |
|---|---|---|---|
| 320 kbps · best | 320k | ~2.4 MB | Music masters, archival, when fidelity matters more than size |
| 256 kbps | 256k | ~1.9 MB | High-quality music distribution, a near-transparent middle ground |
| 192 kbps · default | 192k | ~1.4 MB | Standard distribution — the balanced default for most uploads |
| 128 kbps · podcast | 128k | ~0.94 MB | Podcasts, music-plus-voice where size and reach matter |
| 64 kbps · voice | 64k | ~0.47 MB | Voice-only memos, audiobooks, lectures — smallest file |
What converts and what carries over
What the WAV to MP3 path does and does not change. The tool encodes; it does not resample, remix channels, or normalize loudness.
| Property | Behavior | Notes |
|---|---|---|
| Codec | PCM (WAV) re-encoded to MP3 via libmp3lame | MP3 is lossy — each encode discards some inaudible data; re-encoding an MP3 again would stack loss |
| Sample rate | Passed through from source unchanged | A 48 kHz WAV stays 48 kHz. To change it, use sample-rate-converter |
| Channels | Passed through (mono stays mono, stereo stays stereo) | This tool has no channel control; to fold to mono first use audio-compressor or a channel tool |
| Tags / cover art | Copied to ID3v2.3 (-map_metadata 0, cover as attached_pic) | Edit afterwards with id3-editor |
| Loudness | Untouched — the WAV's levels are encoded as-is | For consistent loudness across files use loudness-normalizer |
Cookbook
Real WAV-to-MP3 decisions, with the bitrate choice that fits each. Commands shown are the FFmpeg the in-browser engine runs under the hood — you only click the bitrate.
A 10-minute stereo WAV to a mailable MP3
A 10-minute 44.1 kHz 16-bit stereo WAV is about 101 MB — too big to email. At 192 kbps it drops to roughly 14 MB, easily attachable, with no audible loss for casual listening.
Source: interview.wav (44.1 kHz / 16-bit / stereo, ~101 MB)
Option: 192 kbps · default
Engine runs:
ffmpeg -i interview.wav -c:a libmp3lame -b:a 192k \
-map_metadata 0 -id3v2_version 3 interview.mp3
Result: interview.mp3 (~14 MB) — ~7x smallerMusic master kept at archival quality
When you want the smallest file that is still essentially transparent for music, pick 320 kbps. It is the highest CBR option and the right default for anything you might keep long-term.
Source: mixdown_final.wav (48 kHz / 24-bit / stereo)
Option: 320 kbps · best
Engine runs:
ffmpeg -i mixdown_final.wav -c:a libmp3lame -b:a 320k \
-map_metadata 0 -id3v2_version 3 mixdown_final.mp3
Result: mixdown_final.mp3 (320 kbps CBR, 48 kHz stereo preserved)Voice memo squeezed for upload
A spoken-word WAV has no music detail to protect, so 64 kbps is plenty and produces a tiny file. Sample rate still passes through; for voice that's usually fine.
Source: voice_note.wav (44.1 kHz / 16-bit / mono, 8 min, ~40 MB)
Option: 64 kbps · voice
Engine runs:
ffmpeg -i voice_note.wav -c:a libmp3lame -b:a 64k \
-map_metadata 0 -id3v2_version 3 voice_note.mp3
Result: voice_note.mp3 (~3.8 MB)Tagged WAV that should stay tagged
If your WAV already carries title/artist/album metadata and embedded artwork, the conversion copies them into the MP3's ID3v2.3 frames automatically — no separate tagging pass needed.
Source: track03.wav (with INFO tags + embedded cover.jpg)
Option: 256 kbps
Engine runs:
ffmpeg -i track03.wav -c:a libmp3lame -b:a 256k \
-map_metadata 0 -map 0:a -map 0:v? -c:v copy \
-disposition:v:0 attached_pic -id3v2_version 3 track03.mp3
Result: track03.mp3 — title/artist/album + cover preservedDeciding 128 vs 192 for a podcast feed
Podcasts are voice-heavy with occasional music. 128 kbps keeps episodes small (matters for listeners on mobile data); 192 kbps gives a little more headroom for intro music. There is no VBR option, so pick the CBR rate that fits your feed's size budget.
Episode WAV: 52 min stereo @ 128 kbps -> ~49 MB (recommended for spoken word) @ 192 kbps -> ~73 MB (more headroom for music beds) Pick per your host's per-episode limit. For a hard size cap, use audio-compressor (/audio-tools/audio-compressor) instead.
Edge cases and what actually happens
WAV file is larger than your tier's size limit
BlockedAudio files are capped per tier: Free 50 MB, Pro 200 MB, Pro-media and Developer 100 GB. A long uncompressed WAV blows past 50 MB quickly (about 5 minutes of CD-quality stereo). If you hit the cap, upgrade, or trim/split the WAV first, then convert each part.
WAV duration exceeds the per-file duration limit
BlockedThere is a separate duration cap: Free 30 minutes, Pro 120 minutes, Pro-media and Developer unlimited. A WAV can be under the byte limit but over the minute limit (a low-sample-rate mono file), so both are checked. Split a long recording before converting if you're on a lower tier.
You expected VBR or a quality slider
By designThis tool encodes constant-bitrate (CBR) MP3 only, selected from five fixed bitrates. There is no VBR mode and no per-file quality slider — the bitrate dropdown is the single control. CBR is the most universally compatible MP3 form, which is why it's the default behavior here.
Output sample rate isn't what you wanted
ExpectedThe encoder passes the source sample rate through unchanged — it does not resample. A 96 kHz WAV becomes a 96 kHz MP3 (which some old players dislike). To force 44.1 or 48 kHz, run the WAV through sample-rate-converter first, then convert to MP3.
Converting an already-lossy file to MP3
Quality cautionIf your 'WAV' was decoded from an MP3 or other lossy source, encoding to MP3 again is a second lossy pass and audible artifacts can stack. The tool will still produce a valid MP3, but you can't recover detail that was already gone. Keep a lossless original where possible.
WAV with 32-bit float samples
SupportedFFmpeg decodes 32-bit float PCM WAV fine and encodes it down to MP3. The float headroom above 0 dBFS is simply baked into the encode; if the float file had peaks above 0 dBFS they may clip. To tame peaks first, use true-peak-limiter or loudness-normalizer.
Multichannel (5.1) WAV dropped in
Channels preservedThe channel layout passes through, so a 5.1 WAV would encode to a multichannel MP3 — but most players expect stereo MP3 and may mishandle it. For predictable playback, downmix to stereo first (the WAV-target conversion tools expose channel options) before converting to MP3.
You dropped a non-WAV audio file
SupportedThe dropzone accepts MP3, WAV, FLAC, M4A, OGG, Opus, or video — FFmpeg decodes them all. But if your source is, say, FLAC, the dedicated flac-to-mp3 tool is the cleaner route. For MP3 you'd want bitrate-changer rather than re-routing through WAV.
Engine fails to load (offline / blocked CDN)
Load errorFFmpeg's WebAssembly bundle downloads on first use. If your network blocks it or you're fully offline before it has cached, the tool can't start. Reload once you have a connection; after the first successful load the engine is cached and conversions run offline.
Corrupt or truncated WAV header
Decode errorIf the WAV header is malformed (incomplete recording, partial download), FFmpeg may fail to decode or produce a short/silent MP3. Re-export the WAV from your DAW or re-download it. A well-formed RIFF/WAVE header is required.
Frequently asked questions
Is my WAV uploaded anywhere?
No. The conversion runs entirely in your browser using FFmpeg 8.1 compiled to WebAssembly. The WAV is read into the page's memory, encoded with libmp3lame, and handed back as a download. Nothing is sent to a server, so even a 150 MB session never crosses the network.
Which bitrate should I pick?
320 kbps for music masters or archival, 192 kbps (the default) for general distribution, 128 kbps for podcasts and music-plus-voice, and 64 kbps for voice-only content where size matters most. Higher bitrate means better fidelity and a larger file. There is no in-between custom value — pick the closest of the five.
Does it support VBR or a quality (V0/V2) setting?
No. This tool produces constant-bitrate (CBR) MP3 only, from the five fixed bitrates. CBR is the most broadly compatible MP3 form. If you need a hard target file size rather than a bitrate, use the audio-compressor tool instead.
Will it change my sample rate or channels?
No — both pass through from the source unchanged. A 48 kHz stereo WAV becomes a 48 kHz stereo MP3. To change sample rate, run it through sample-rate-converter first; this converter has no resample or channel control.
Are my ID3 tags and cover art kept?
Yes. The encoder copies metadata with -map_metadata 0 and re-attaches any embedded cover image as attached_pic, writing ID3v2.3. A tagged WAV stays tagged. To add or fix tags afterwards, use the id3-editor tool.
Will I lose quality converting WAV to MP3?
MP3 is a lossy format, so some data is discarded during encoding. At 192-320 kbps the loss is inaudible to most listeners on most material. The original WAV is untouched on your disk — keep it as the lossless master and treat the MP3 as the distribution copy.
How big a WAV can I convert?
Free tier allows up to 50 MB and 30 minutes per file; Pro raises that to 200 MB and 120 minutes with batch support; Pro-media and Developer allow up to 100 GB with unlimited duration. Both size and duration are checked, so a long low-rate WAV can hit the minute cap before the byte cap.
Can I convert several WAVs at once?
This specific tool processes one file at a time (no multi-file dropzone). Pro and higher tiers support batch limits of 10+ files across audio tools, but the WAV-to-MP3 page itself is single-file — drop, convert, repeat. For merging several into one file, see audio-merger.
Why is the converted MP3 still fairly large?
File size is bitrate times duration. A long recording at 320 kbps is still big. Drop to 192 or 128 kbps for a smaller file, or if you need a specific size ceiling (for an email or upload limit), use audio-compressor, which targets a megabyte budget rather than a bitrate.
Does it work offline?
After the first load it does. The FFmpeg WebAssembly engine downloads once (a few MB) and is cached by the browser; subsequent conversions run with no network. The very first conversion needs a connection to fetch the engine.
What's the difference between this and the audio-compressor?
This tool lets you pick an MP3 bitrate. The audio-compressor tool targets a file size in megabytes and works backward to the bitrate that fits — useful for hard upload caps. Use this for predictable quality, the compressor for predictable size.
Can I go back from MP3 to WAV later?
You can decode an MP3 back to WAV with mp3-to-wav, but it will not restore the data MP3 dropped — the result is an uncompressed file of the lossy audio, useful for editing without further generational loss, not a recovery of the original WAV's fidelity.
Privacy first
Every JAD Audio tool runs entirely in your browser via FFmpeg (WebAssembly) and RNNoise. Your audio files never leave your device — verified by zero outbound network requests during processing.