How to convert WAV to MP3

Four methods, the bitrate to choose, and the one mistake worth avoiding. If you only need the file, the browser converter takes about ten seconds.

Method 1: in your browser

1. Open the WAV to MP3 converter and drop your files onto it.
2. Pick a bitrate. 192 kbps is the default and suits most things. Open Output settings to move to 320 kbps for music, or down to 96 kbps for speech.
3. Download. Preview it in the built-in player first if you want to be sure, then save the file or grab the whole batch as a ZIP.

The encoder is LAME, the same one Audacity and ffmpeg use, so the output is not a compromise version. Because everything runs locally, an unreleased master never leaves your machine.

Method 2: Audacity

Worth using when you want to trim, fade, or normalise before exporting.

1. Open the WAV by dragging it into the window.
2. Choose File, then Export Audio. Older versions use File, Export, Export as MP3.
3. Set the format to MP3 and choose your bit rate mode. Preset or Variable gives better quality per megabyte. Constant is the safe choice if something downstream demands a fixed bitrate.
4. Fill in the metadata. Unlike WAV, MP3 tags are well supported, so artist and title entered here will show up in every player.
5. Export. Modern Audacity includes the MP3 encoder, so there is no separate LAME download to hunt for any more.

Method 3: ffmpeg

The fastest option for large batches. A basic conversion at 192 kbps:

ffmpeg -i input.wav -codec:a libmp3lame -b:a 192k output.mp3

For 320 kbps constant bitrate, which is the highest MP3 supports:

ffmpeg -i input.wav -codec:a libmp3lame -b:a 320k output.mp3

For variable bitrate, use the quality scale instead. -q:a 0 is the highest setting and averages somewhere around 220 to 260 kbps on music, though the exact figure moves with the material:

ffmpeg -i input.wav -codec:a libmp3lame -q:a 0 output.mp3

Converting a folder on macOS or Linux:

for f in *.wav; do ffmpeg -i "$f" -codec:a libmp3lame -b:a 192k "${f%.wav}.mp3"; done

Mono speech at a small size: add -ac 1 and drop the bitrate to -b:a 96k.

Method 4: Apple Music or iTunes

1. Open Import Settings. Music, Settings, Files, Import Settings on macOS. Edit, Preferences, General, Import Settings in iTunes for Windows.
2. Set Import Using to MP3 Encoder and choose a quality setting. Higher Quality is 160 kbps, Custom lets you reach 320.
3. Add the WAV to your library, select it, then File, Convert, Create MP3 Version.
4. Put the import setting back to AAC afterwards if that is what you normally use.

Choosing a bitrate

One habit is worth building: always encode from the WAV, never from another MP3. Each pass through the encoder adds damage on top of the last, and the second generation is where it starts to become audible. More on what bitrate changes.

CBR or VBR

Constant bitrate spends the same number of bits on every second of audio, including the quiet ones. Variable bitrate lets the encoder spend more where the music is dense and less where it is sparse, which means better quality for the same file size.

VBR is the better default. Reach for CBR when a broadcast system, a phone system, or an older hardware player specifies it, or when something needs to predict the file size in advance. The converter on this site uses constant bitrate, which keeps behaviour predictable across every browser.

When something goes wrong

The MP3 sounds dull compared to the WAV. Try a higher bitrate. Cymbals, applause, and solo piano expose low bitrate encoding faster than anything else.

Encoding feels slow. MP3 encoding is genuinely more work than writing a WAV, because the encoder analyses the audio to decide what to discard. A four minute track takes a few seconds. An hour long recording takes a while, and ffmpeg will be quicker than a browser for that.

A 32-bit float WAV will not convert. Some older tools only read 16-bit PCM. Re-export the WAV as 16-bit first. The browser converter here handles 16, 24, and 32-bit float without complaint.

The file was already an MP3. If your WAV was made by converting an MP3, going back to MP3 is a second generation encode and will sound worse than the original did. Find the first MP3 and use that instead where you can.

Tags are missing. WAV carries metadata poorly, so artist and title often do not survive the trip. Add them after converting, in Audacity, in your music player, or with a tag editor.

Common questions